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gstreamer/subprojects/gst-plugins-bad/sys/androidmedia/gstamcaudiodec.c

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/*
* Initially based on gst-omx/omx/gstomxvideodec.c
*
* Copyright (C) 2011, Hewlett-Packard Development Company, L.P.
* Author: Sebastian Dröge <sebastian.droege@collabora.co.uk>, Collabora Ltd.
*
* Copyright (C) 2012, Collabora Ltd.
* Author: Sebastian Dröge <sebastian.droege@collabora.co.uk>
*
* Copyright (C) 2015, Sebastian Dröge <sebastian@centricular.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation
* version 2.1 of the License.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst.h>
#include <gst/audio/audio.h>
#include <string.h>
#ifdef HAVE_ORC
#include <orc/orc.h>
#else
#define orc_memcpy memcpy
#endif
#include "gstamcaudiodec.h"
#include "gstamc-constants.h"
GST_DEBUG_CATEGORY_STATIC (gst_amc_audio_dec_debug_category);
#define GST_CAT_DEFAULT gst_amc_audio_dec_debug_category
#define GST_AUDIO_DECODER_ERROR_FROM_ERROR(el, err) G_STMT_START { \
gchar *__dbg = g_strdup (err->message); \
GstAudioDecoder *__dec = GST_AUDIO_DECODER (el); \
GST_WARNING_OBJECT (el, "error: %s", __dbg); \
_gst_audio_decoder_error (__dec, 1, \
err->domain, err->code, \
NULL, __dbg, __FILE__, GST_FUNCTION, __LINE__); \
g_clear_error (&err); \
} G_STMT_END
/* prototypes */
static void gst_amc_audio_dec_finalize (GObject * object);
static GstStateChangeReturn
gst_amc_audio_dec_change_state (GstElement * element,
GstStateChange transition);
static gboolean gst_amc_audio_dec_open (GstAudioDecoder * decoder);
static gboolean gst_amc_audio_dec_close (GstAudioDecoder * decoder);
static gboolean gst_amc_audio_dec_start (GstAudioDecoder * decoder);
static gboolean gst_amc_audio_dec_stop (GstAudioDecoder * decoder);
static gboolean gst_amc_audio_dec_set_format (GstAudioDecoder * decoder,
GstCaps * caps);
static void gst_amc_audio_dec_flush (GstAudioDecoder * decoder, gboolean hard);
static GstFlowReturn gst_amc_audio_dec_handle_frame (GstAudioDecoder * decoder,
GstBuffer * buffer);
static GstFlowReturn gst_amc_audio_dec_drain (GstAmcAudioDec * self);
enum
{
PROP_0
};
/* class initialization */
static void gst_amc_audio_dec_class_init (GstAmcAudioDecClass * klass);
static void gst_amc_audio_dec_init (GstAmcAudioDec * self);
static void gst_amc_audio_dec_base_init (gpointer g_class);
static GstAudioDecoderClass *parent_class = NULL;
GType
gst_amc_audio_dec_get_type (void)
{
static gsize type = 0;
if (g_once_init_enter (&type)) {
GType _type;
static const GTypeInfo info = {
sizeof (GstAmcAudioDecClass),
gst_amc_audio_dec_base_init,
NULL,
(GClassInitFunc) gst_amc_audio_dec_class_init,
NULL,
NULL,
sizeof (GstAmcAudioDec),
0,
(GInstanceInitFunc) gst_amc_audio_dec_init,
NULL
};
_type = g_type_register_static (GST_TYPE_AUDIO_DECODER, "GstAmcAudioDec",
&info, 0);
GST_DEBUG_CATEGORY_INIT (gst_amc_audio_dec_debug_category, "amcaudiodec", 0,
"Android MediaCodec audio decoder");
g_once_init_leave (&type, _type);
}
return type;
}
static const gchar *
caps_to_mime (GstCaps * caps)
{
GstStructure *s;
const gchar *name;
s = gst_caps_get_structure (caps, 0);
if (!s)
return NULL;
name = gst_structure_get_name (s);
if (strcmp (name, "audio/mpeg") == 0) {
gint mpegversion;
if (!gst_structure_get_int (s, "mpegversion", &mpegversion))
return NULL;
if (mpegversion == 1) {
gint layer;
if (!gst_structure_get_int (s, "layer", &layer) || layer == 3)
return "audio/mpeg";
else if (layer == 2)
return "audio/mpeg-L2";
} else if (mpegversion == 2 || mpegversion == 4) {
return "audio/mp4a-latm";
}
} else if (strcmp (name, "audio/AMR") == 0) {
return "audio/3gpp";
} else if (strcmp (name, "audio/AMR-WB") == 0) {
return "audio/amr-wb";
} else if (strcmp (name, "audio/x-alaw") == 0) {
return "audio/g711-alaw";
} else if (strcmp (name, "audio/x-mulaw") == 0) {
return "audio/g711-mlaw";
} else if (strcmp (name, "audio/x-vorbis") == 0) {
return "audio/vorbis";
} else if (strcmp (name, "audio/x-opus") == 0) {
return "audio/opus";
}
return NULL;
}
static void
gst_amc_audio_dec_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
GstAmcAudioDecClass *amcaudiodec_class = GST_AMC_AUDIO_DEC_CLASS (g_class);
const GstAmcCodecInfo *codec_info;
GstPadTemplate *templ;
GstCaps *sink_caps, *src_caps;
gchar *longname;
codec_info =
g_type_get_qdata (G_TYPE_FROM_CLASS (g_class), gst_amc_codec_info_quark);
/* This happens for the base class and abstract subclasses */
if (!codec_info)
return;
amcaudiodec_class->codec_info = codec_info;
gst_amc_codec_info_to_caps (codec_info, &sink_caps, &src_caps);
/* Add pad templates */
templ =
gst_pad_template_new ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, sink_caps);
gst_element_class_add_pad_template (element_class, templ);
gst_caps_unref (sink_caps);
templ = gst_pad_template_new ("src", GST_PAD_SRC, GST_PAD_ALWAYS, src_caps);
gst_element_class_add_pad_template (element_class, templ);
gst_caps_unref (src_caps);
longname = g_strdup_printf ("Android MediaCodec %s", codec_info->name);
gst_element_class_set_metadata (element_class,
codec_info->name,
"Codec/Decoder/Audio",
longname, "Sebastian Dröge <sebastian.droege@collabora.co.uk>");
g_free (longname);
}
static void
gst_amc_audio_dec_class_init (GstAmcAudioDecClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstAudioDecoderClass *audiodec_class = GST_AUDIO_DECODER_CLASS (klass);
parent_class = g_type_class_peek_parent (klass);
gobject_class->finalize = gst_amc_audio_dec_finalize;
element_class->change_state =
GST_DEBUG_FUNCPTR (gst_amc_audio_dec_change_state);
audiodec_class->start = GST_DEBUG_FUNCPTR (gst_amc_audio_dec_start);
audiodec_class->stop = GST_DEBUG_FUNCPTR (gst_amc_audio_dec_stop);
audiodec_class->open = GST_DEBUG_FUNCPTR (gst_amc_audio_dec_open);
audiodec_class->close = GST_DEBUG_FUNCPTR (gst_amc_audio_dec_close);
audiodec_class->flush = GST_DEBUG_FUNCPTR (gst_amc_audio_dec_flush);
audiodec_class->set_format = GST_DEBUG_FUNCPTR (gst_amc_audio_dec_set_format);
audiodec_class->handle_frame =
GST_DEBUG_FUNCPTR (gst_amc_audio_dec_handle_frame);
}
static void
gst_amc_audio_dec_init (GstAmcAudioDec * self)
{
gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (self), TRUE);
gst_audio_decoder_set_drainable (GST_AUDIO_DECODER (self), TRUE);
g_mutex_init (&self->drain_lock);
g_cond_init (&self->drain_cond);
self->output_adapter = gst_adapter_new ();
}
static gboolean
gst_amc_audio_dec_open (GstAudioDecoder * decoder)
{
GstAmcAudioDec *self = GST_AMC_AUDIO_DEC (decoder);
GstAmcAudioDecClass *klass = GST_AMC_AUDIO_DEC_GET_CLASS (self);
GError *err = NULL;
GST_DEBUG_OBJECT (self, "Opening decoder");
self->codec = gst_amc_codec_new (klass->codec_info->name, FALSE, &err);
if (!self->codec) {
GST_ELEMENT_ERROR_FROM_ERROR (self, err);
return FALSE;
}
self->started = FALSE;
self->flushing = TRUE;
GST_DEBUG_OBJECT (self, "Opened decoder");
return TRUE;
}
static gboolean
gst_amc_audio_dec_close (GstAudioDecoder * decoder)
{
GstAmcAudioDec *self = GST_AMC_AUDIO_DEC (decoder);
GST_DEBUG_OBJECT (self, "Closing decoder");
if (self->codec) {
GError *err = NULL;
gst_amc_codec_release (self->codec, &err);
if (err)
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
gst_amc_codec_free (self->codec);
}
self->codec = NULL;
self->started = FALSE;
self->flushing = TRUE;
GST_DEBUG_OBJECT (self, "Closed decoder");
return TRUE;
}
static void
gst_amc_audio_dec_finalize (GObject * object)
{
GstAmcAudioDec *self = GST_AMC_AUDIO_DEC (object);
if (self->output_adapter)
gst_object_unref (self->output_adapter);
self->output_adapter = NULL;
g_mutex_clear (&self->drain_lock);
g_cond_clear (&self->drain_cond);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static GstStateChangeReturn
gst_amc_audio_dec_change_state (GstElement * element, GstStateChange transition)
{
GstAmcAudioDec *self;
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
GError *err = NULL;
g_return_val_if_fail (GST_IS_AMC_AUDIO_DEC (element),
GST_STATE_CHANGE_FAILURE);
self = GST_AMC_AUDIO_DEC (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
self->downstream_flow_ret = GST_FLOW_OK;
self->draining = FALSE;
self->started = FALSE;
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
self->flushing = TRUE;
gst_amc_codec_flush (self->codec, &err);
if (err)
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
g_mutex_lock (&self->drain_lock);
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
g_mutex_unlock (&self->drain_lock);
break;
default:
break;
}
if (ret == GST_STATE_CHANGE_FAILURE)
return ret;
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
if (ret == GST_STATE_CHANGE_FAILURE)
return ret;
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
self->downstream_flow_ret = GST_FLOW_FLUSHING;
self->started = FALSE;
break;
case GST_STATE_CHANGE_READY_TO_NULL:
break;
default:
break;
}
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 */
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 void
gst_amc_audio_dec_loop (GstAmcAudioDec * self)
{
GstFlowReturn flow_ret = GST_FLOW_OK;
gboolean is_eos;
GstAmcBuffer *buf;
GstAmcBufferInfo buffer_info;
gint idx;
GError *err = NULL;
GST_AUDIO_DECODER_STREAM_LOCK (self);
retry:
/*if (self->input_caps_changed) {
idx = INFO_OUTPUT_FORMAT_CHANGED;
} else { */
GST_DEBUG_OBJECT (self, "Waiting for available output buffer");
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
/* Wait at most 100ms here, some codecs don't fail dequeueing if
* the codec is flushing, causing deadlocks during shutdown */
idx =
gst_amc_codec_dequeue_output_buffer (self->codec, &buffer_info, 100000,
&err);
GST_AUDIO_DECODER_STREAM_LOCK (self);
/*} */
if (idx < 0) {
if (self->flushing) {
g_clear_error (&err);
goto flushing;
}
switch (idx) {
case INFO_OUTPUT_BUFFERS_CHANGED:
/* Handled internally */
g_assert_not_reached ();
break;
case INFO_OUTPUT_FORMAT_CHANGED:{
GstAmcFormat *format;
gchar *format_string;
GST_DEBUG_OBJECT (self, "Output format has changed");
format = gst_amc_codec_get_output_format (self->codec, &err);
if (!format)
goto format_error;
format_string = gst_amc_format_to_string (format, &err);
if (err) {
gst_amc_format_free (format);
goto format_error;
}
GST_DEBUG_OBJECT (self, "Got new output format: %s", format_string);
g_free (format_string);
if (!gst_amc_audio_dec_set_src_caps (self, format)) {
gst_amc_format_free (format);
goto format_error;
}
gst_amc_format_free (format);
goto retry;
}
case INFO_TRY_AGAIN_LATER:
GST_DEBUG_OBJECT (self, "Dequeueing output buffer timed out");
goto retry;
case G_MININT:
GST_ERROR_OBJECT (self, "Failure dequeueing output buffer");
goto dequeue_error;
default:
g_assert_not_reached ();
break;
}
goto retry;
}
GST_DEBUG_OBJECT (self,
"Got output buffer at index %d: offset %d size %d time %" G_GINT64_FORMAT
" flags 0x%08x", idx, buffer_info.offset, buffer_info.size,
buffer_info.presentation_time_us, buffer_info.flags);
is_eos = ! !(buffer_info.flags & BUFFER_FLAG_END_OF_STREAM);
buf = gst_amc_codec_get_output_buffer (self->codec, idx, &err);
if (err) {
if (self->flushing) {
g_clear_error (&err);
goto flushing;
}
goto failed_to_get_output_buffer;
} else if (!buf) {
goto got_null_output_buffer;
}
if (buffer_info.size > 0) {
GstBuffer *outbuf;
GstMapInfo minfo;
/* This sometimes happens at EOS or if the input is not properly framed,
* let's handle it gracefully by allocating a new buffer for the current
* caps and filling it
*/
if (buffer_info.size % self->info.bpf != 0)
goto invalid_buffer_size;
outbuf =
gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER (self),
buffer_info.size);
if (!outbuf)
goto failed_allocate;
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->data + buffer_info.offset);
n_samples = buffer_info.size / 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 {
orc_memcpy (minfo.data, buf->data + buffer_info.offset, buffer_info.size);
}
gst_buffer_unmap (outbuf, &minfo);
if (self->spf != -1) {
gst_adapter_push (self->output_adapter, outbuf);
} else {
flow_ret =
gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (self), outbuf, 1);
}
}
gst_amc_buffer_free (buf);
buf = NULL;
if (self->spf != -1) {
GstBuffer *outbuf;
guint avail = gst_adapter_available (self->output_adapter);
guint nframes;
/* On EOS we take the complete adapter content, no matter
* if it is a multiple of the codec frame size or not.
* Otherwise we take a multiple of codec frames and push
* them downstream
*/
avail /= self->info.bpf;
if (!is_eos) {
nframes = avail / self->spf;
avail = nframes * self->spf;
} else {
nframes = (avail + self->spf - 1) / self->spf;
}
avail *= self->info.bpf;
if (avail > 0) {
outbuf = gst_adapter_take_buffer (self->output_adapter, avail);
flow_ret =
gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (self), outbuf,
nframes);
}
}
if (!gst_amc_codec_release_output_buffer (self->codec, idx, FALSE, &err)) {
if (self->flushing) {
g_clear_error (&err);
goto flushing;
}
goto failed_release;
}
if (is_eos || flow_ret == GST_FLOW_EOS) {
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
g_mutex_lock (&self->drain_lock);
if (self->draining) {
GST_DEBUG_OBJECT (self, "Drained");
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
} else if (flow_ret == GST_FLOW_OK) {
GST_DEBUG_OBJECT (self, "Component signalled EOS");
flow_ret = GST_FLOW_EOS;
}
g_mutex_unlock (&self->drain_lock);
GST_AUDIO_DECODER_STREAM_LOCK (self);
} else {
GST_DEBUG_OBJECT (self, "Finished frame: %s", gst_flow_get_name (flow_ret));
}
self->downstream_flow_ret = flow_ret;
if (flow_ret != GST_FLOW_OK)
goto flow_error;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
return;
dequeue_error:
{
GST_ELEMENT_ERROR_FROM_ERROR (self, 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;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
g_mutex_lock (&self->drain_lock);
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
g_mutex_unlock (&self->drain_lock);
return;
}
format_error:
{
if (err)
GST_ELEMENT_ERROR_FROM_ERROR (self, err);
else
GST_ELEMENT_ERROR (self, LIBRARY, FAILED, (NULL),
("Failed to handle format"));
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;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
g_mutex_lock (&self->drain_lock);
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
g_mutex_unlock (&self->drain_lock);
return;
}
failed_release:
{
GST_AUDIO_DECODER_ERROR_FROM_ERROR (self, 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;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
g_mutex_lock (&self->drain_lock);
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
g_mutex_unlock (&self->drain_lock);
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;
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));
} else if (flow_ret < GST_FLOW_EOS) {
GST_ELEMENT_FLOW_ERROR (self, 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));
} else if (flow_ret == GST_FLOW_FLUSHING) {
GST_DEBUG_OBJECT (self, "Flushing -- stopping task");
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
}
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
g_mutex_lock (&self->drain_lock);
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
g_mutex_unlock (&self->drain_lock);
return;
}
failed_to_get_output_buffer:
{
GST_AUDIO_DECODER_ERROR_FROM_ERROR (self, 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;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
g_mutex_lock (&self->drain_lock);
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
g_mutex_unlock (&self->drain_lock);
return;
}
got_null_output_buffer:
{
GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL),
("Got no output 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_ERROR;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
g_mutex_lock (&self->drain_lock);
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
g_mutex_unlock (&self->drain_lock);
return;
}
invalid_buffer_size:
{
GST_ELEMENT_ERROR (self, LIBRARY, FAILED, (NULL),
("Invalid buffer size %u (bfp %d)", buffer_info.size, self->info.bpf));
gst_amc_codec_release_output_buffer (self->codec, idx, FALSE, &err);
if (err && !self->flushing)
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
g_clear_error (&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;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
g_mutex_lock (&self->drain_lock);
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
g_mutex_unlock (&self->drain_lock);
return;
}
failed_allocate:
{
GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL),
("Failed to allocate output buffer"));
gst_amc_codec_release_output_buffer (self->codec, idx, FALSE, &err);
if (err && !self->flushing)
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
g_clear_error (&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;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
g_mutex_lock (&self->drain_lock);
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
g_mutex_unlock (&self->drain_lock);
return;
}
}
static gboolean
gst_amc_audio_dec_start (GstAudioDecoder * decoder)
{
GstAmcAudioDec *self;
self = GST_AMC_AUDIO_DEC (decoder);
self->last_upstream_ts = 0;
self->drained = TRUE;
self->downstream_flow_ret = GST_FLOW_OK;
self->started = FALSE;
self->flushing = TRUE;
return TRUE;
}
static gboolean
gst_amc_audio_dec_stop (GstAudioDecoder * decoder)
{
GstAmcAudioDec *self;
GError *err = NULL;
self = GST_AMC_AUDIO_DEC (decoder);
GST_DEBUG_OBJECT (self, "Stopping decoder");
self->flushing = TRUE;
if (self->started) {
gst_amc_codec_flush (self->codec, &err);
if (err)
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
gst_amc_codec_stop (self->codec, &err);
if (err)
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
self->started = FALSE;
}
gst_pad_stop_task (GST_AUDIO_DECODER_SRC_PAD (decoder));
memset (self->positions, 0, sizeof (self->positions));
gst_adapter_flush (self->output_adapter,
gst_adapter_available (self->output_adapter));
g_list_foreach (self->codec_datas, (GFunc) g_free, NULL);
g_list_free (self->codec_datas);
self->codec_datas = NULL;
self->downstream_flow_ret = GST_FLOW_FLUSHING;
self->drained = TRUE;
g_mutex_lock (&self->drain_lock);
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
g_mutex_unlock (&self->drain_lock);
GST_DEBUG_OBJECT (self, "Stopped decoder");
return TRUE;
}
static gboolean
gst_amc_audio_dec_set_format (GstAudioDecoder * decoder, GstCaps * caps)
{
GstAmcAudioDec *self;
GstStructure *s;
GstAmcFormat *format;
const gchar *mime;
gboolean is_format_change = FALSE;
gboolean needs_disable = FALSE;
gchar *format_string;
gint rate, channels;
GError *err = NULL;
self = GST_AMC_AUDIO_DEC (decoder);
GST_DEBUG_OBJECT (self, "Setting new caps %" GST_PTR_FORMAT, caps);
/* Check if the caps change is a real format change or if only irrelevant
* parts of the caps have changed or nothing at all.
*/
is_format_change |= (!self->input_caps
|| !gst_caps_is_equal (self->input_caps, caps));
needs_disable = self->started;
/* If the component is not started and a real format change happens
* we have to restart the component. If no real format change
* happened we can just exit here.
*/
if (needs_disable && !is_format_change) {
/* Framerate or something minor changed */
self->input_caps_changed = TRUE;
GST_DEBUG_OBJECT (self,
"Already running and caps did not change the format");
return TRUE;
}
if (needs_disable && is_format_change) {
gst_amc_audio_dec_drain (self);
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
gst_amc_audio_dec_stop (GST_AUDIO_DECODER (self));
GST_AUDIO_DECODER_STREAM_LOCK (self);
gst_amc_audio_dec_close (GST_AUDIO_DECODER (self));
if (!gst_amc_audio_dec_open (GST_AUDIO_DECODER (self))) {
GST_ERROR_OBJECT (self, "Failed to open codec again");
return FALSE;
}
if (!gst_amc_audio_dec_start (GST_AUDIO_DECODER (self))) {
GST_ERROR_OBJECT (self, "Failed to start codec again");
}
}
/* srcpad task is not running at this point */
mime = caps_to_mime (caps);
if (!mime) {
GST_ERROR_OBJECT (self, "Failed to convert caps to mime");
return FALSE;
}
s = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_int (s, "rate", &rate) ||
!gst_structure_get_int (s, "channels", &channels)) {
GST_ERROR_OBJECT (self, "Failed to get rate/channels");
return FALSE;
}
format = gst_amc_format_new_audio (mime, rate, channels, &err);
if (!format) {
GST_ELEMENT_ERROR_FROM_ERROR (self, err);
return FALSE;
}
if (gst_structure_has_name (s, "audio/mpeg")) {
gint mpegversion;
const gchar *stream_format;
if (!gst_structure_get_int (s, "mpegversion", &mpegversion))
mpegversion = -1;
stream_format = gst_structure_get_string (s, "stream-format");
if (mpegversion == 4 && g_strcmp0 (stream_format, "adts") == 0) {
gst_amc_format_set_int (format, "is-adts", 1, &err);
if (err)
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
}
}
/* FIXME: These buffers needs to be valid until the codec is stopped again */
g_list_foreach (self->codec_datas, (GFunc) gst_buffer_unref, NULL);
g_list_free (self->codec_datas);
self->codec_datas = NULL;
if (gst_structure_has_field (s, "codec_data")) {
const GValue *h = gst_structure_get_value (s, "codec_data");
GstBuffer *codec_data = gst_value_get_buffer (h);
GstMapInfo minfo;
guint8 *data;
gst_buffer_map (codec_data, &minfo, GST_MAP_READ);
data = g_memdup2 (minfo.data, minfo.size);
self->codec_datas = g_list_prepend (self->codec_datas, data);
gst_amc_format_set_buffer (format, "csd-0", data, minfo.size, &err);
if (err)
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
gst_buffer_unmap (codec_data, &minfo);
} else if (gst_structure_has_field (s, "streamheader")) {
const GValue *sh = gst_structure_get_value (s, "streamheader");
gint nsheaders = gst_value_array_get_size (sh);
GstBuffer *buf;
const GValue *h;
gint i, j;
gchar *fname;
GstMapInfo minfo;
guint8 *data;
for (i = 0, j = 0; i < nsheaders; i++) {
h = gst_value_array_get_value (sh, i);
buf = gst_value_get_buffer (h);
if (strcmp (mime, "audio/vorbis") == 0) {
guint8 header_type;
gst_buffer_extract (buf, 0, &header_type, 1);
/* Only use the identification and setup packets */
if (header_type != 0x01 && header_type != 0x05)
continue;
}
fname = g_strdup_printf ("csd-%d", j);
gst_buffer_map (buf, &minfo, GST_MAP_READ);
data = g_memdup2 (minfo.data, minfo.size);
self->codec_datas = g_list_prepend (self->codec_datas, data);
gst_amc_format_set_buffer (format, fname, data, minfo.size, &err);
if (err)
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
gst_buffer_unmap (buf, &minfo);
g_free (fname);
j++;
}
}
format_string = gst_amc_format_to_string (format, &err);
if (err)
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
GST_DEBUG_OBJECT (self, "Configuring codec with format: %s",
GST_STR_NULL (format_string));
g_free (format_string);
if (!gst_amc_codec_configure (self->codec, format, NULL, &err)) {
GST_ERROR_OBJECT (self, "Failed to configure codec");
GST_ELEMENT_ERROR_FROM_ERROR (self, err);
return FALSE;
}
gst_amc_format_free (format);
if (!gst_amc_codec_start (self->codec, &err)) {
GST_ERROR_OBJECT (self, "Failed to start codec");
GST_ELEMENT_ERROR_FROM_ERROR (self, err);
return FALSE;
}
self->spf = -1;
/* TODO: Implement for other codecs too */
if (gst_structure_has_name (s, "audio/mpeg")) {
gint mpegversion = -1;
gst_structure_get_int (s, "mpegversion", &mpegversion);
if (mpegversion == 1) {
gint layer = -1, mpegaudioversion = -1;
gst_structure_get_int (s, "layer", &layer);
gst_structure_get_int (s, "mpegaudioversion", &mpegaudioversion);
if (layer == 1)
self->spf = 384;
else if (layer == 2)
self->spf = 1152;
else if (layer == 3 && mpegaudioversion != -1)
self->spf = (mpegaudioversion == 1 ? 1152 : 576);
}
}
self->started = TRUE;
self->input_caps_changed = TRUE;
/* Start the srcpad loop again */
self->flushing = FALSE;
self->downstream_flow_ret = GST_FLOW_OK;
gst_pad_start_task (GST_AUDIO_DECODER_SRC_PAD (self),
(GstTaskFunction) gst_amc_audio_dec_loop, decoder, NULL);
return TRUE;
}
static void
gst_amc_audio_dec_flush (GstAudioDecoder * decoder, gboolean hard)
{
GstAmcAudioDec *self;
GError *err = NULL;
self = GST_AMC_AUDIO_DEC (decoder);
GST_DEBUG_OBJECT (self, "Resetting decoder");
if (!self->started) {
GST_DEBUG_OBJECT (self, "Codec not started yet");
return;
}
self->flushing = TRUE;
/* Wait until the srcpad loop is finished,
* unlock GST_AUDIO_DECODER_STREAM_LOCK to prevent deadlocks
* caused by using this lock from inside the loop function */
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
GST_PAD_STREAM_LOCK (GST_AUDIO_DECODER_SRC_PAD (self));
GST_PAD_STREAM_UNLOCK (GST_AUDIO_DECODER_SRC_PAD (self));
GST_AUDIO_DECODER_STREAM_LOCK (self);
gst_amc_codec_flush (self->codec, &err);
if (err)
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
gst_adapter_flush (self->output_adapter,
gst_adapter_available (self->output_adapter));
self->flushing = FALSE;
/* Start the srcpad loop again */
self->last_upstream_ts = 0;
self->drained = TRUE;
self->downstream_flow_ret = GST_FLOW_OK;
gst_pad_start_task (GST_AUDIO_DECODER_SRC_PAD (self),
(GstTaskFunction) gst_amc_audio_dec_loop, decoder, NULL);
GST_DEBUG_OBJECT (self, "Reset decoder");
}
static GstFlowReturn
gst_amc_audio_dec_handle_frame (GstAudioDecoder * decoder, GstBuffer * inbuf)
{
GstAmcAudioDec *self;
gint idx;
GstAmcBuffer *buf;
GstAmcBufferInfo buffer_info;
guint offset = 0;
GstClockTime timestamp, duration, timestamp_offset = 0;
GstMapInfo minfo;
GError *err = NULL;
memset (&minfo, 0, sizeof (minfo));
self = GST_AMC_AUDIO_DEC (decoder);
GST_DEBUG_OBJECT (self, "Handling frame");
/* Make sure to keep a reference to the input here,
* it can be unreffed from the other thread if
* finish_frame() is called */
if (inbuf)
inbuf = gst_buffer_ref (inbuf);
if (!self->started) {
GST_ERROR_OBJECT (self, "Codec not started yet");
if (inbuf)
gst_buffer_unref (inbuf);
return GST_FLOW_NOT_NEGOTIATED;
}
if (self->flushing)
goto flushing;
if (self->downstream_flow_ret != GST_FLOW_OK)
goto downstream_error;
if (!inbuf)
return gst_amc_audio_dec_drain (self);
timestamp = GST_BUFFER_PTS (inbuf);
duration = GST_BUFFER_DURATION (inbuf);
gst_buffer_map (inbuf, &minfo, GST_MAP_READ);
while (offset < minfo.size) {
/* Make sure to release the base class stream lock, otherwise
* _loop() can't call _finish_frame() and we might block forever
* because no input buffers are released */
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
/* Wait at most 100ms here, some codecs don't fail dequeueing if
* the codec is flushing, causing deadlocks during shutdown */
idx = gst_amc_codec_dequeue_input_buffer (self->codec, 100000, &err);
GST_AUDIO_DECODER_STREAM_LOCK (self);
if (idx < 0) {
if (self->flushing || self->downstream_flow_ret == GST_FLOW_FLUSHING) {
g_clear_error (&err);
goto flushing;
}
switch (idx) {
case INFO_TRY_AGAIN_LATER:
GST_DEBUG_OBJECT (self, "Dequeueing input buffer timed out");
continue; /* next try */
break;
case G_MININT:
GST_ERROR_OBJECT (self, "Failed to dequeue input buffer");
goto dequeue_error;
default:
g_assert_not_reached ();
break;
}
continue;
}
if (self->flushing) {
memset (&buffer_info, 0, sizeof (buffer_info));
gst_amc_codec_queue_input_buffer (self->codec, idx, &buffer_info, NULL);
goto flushing;
}
if (self->downstream_flow_ret != GST_FLOW_OK) {
memset (&buffer_info, 0, sizeof (buffer_info));
gst_amc_codec_queue_input_buffer (self->codec, idx, &buffer_info, &err);
if (err && !self->flushing)
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
g_clear_error (&err);
goto downstream_error;
}
/* Now handle the frame */
/* Copy the buffer content in chunks of size as requested
* by the port */
buf = gst_amc_codec_get_input_buffer (self->codec, idx, &err);
if (err)
goto failed_to_get_input_buffer;
else if (!buf)
goto got_null_input_buffer;
memset (&buffer_info, 0, sizeof (buffer_info));
buffer_info.offset = 0;
buffer_info.size = MIN (minfo.size - offset, buf->size);
gst_amc_buffer_set_position_and_limit (buf, NULL, buffer_info.offset,
buffer_info.size);
orc_memcpy (buf->data, minfo.data + offset, buffer_info.size);
gst_amc_buffer_free (buf);
buf = NULL;
/* Interpolate timestamps if we're passing the buffer
* in multiple chunks */
if (offset != 0 && duration != GST_CLOCK_TIME_NONE) {
timestamp_offset = gst_util_uint64_scale (offset, duration, minfo.size);
}
if (timestamp != GST_CLOCK_TIME_NONE) {
buffer_info.presentation_time_us =
gst_util_uint64_scale (timestamp + timestamp_offset, 1, GST_USECOND);
self->last_upstream_ts = timestamp + timestamp_offset;
}
if (duration != GST_CLOCK_TIME_NONE)
self->last_upstream_ts += duration;
if (offset == 0) {
if (!GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_DELTA_UNIT))
buffer_info.flags |= BUFFER_FLAG_SYNC_FRAME;
}
offset += buffer_info.size;
GST_DEBUG_OBJECT (self,
"Queueing buffer %d: size %d time %" G_GINT64_FORMAT " flags 0x%08x",
idx, buffer_info.size, buffer_info.presentation_time_us,
buffer_info.flags);
if (!gst_amc_codec_queue_input_buffer (self->codec, idx, &buffer_info,
&err)) {
if (self->flushing) {
g_clear_error (&err);
goto flushing;
}
goto queue_error;
}
self->drained = FALSE;
}
gst_buffer_unmap (inbuf, &minfo);
gst_buffer_unref (inbuf);
return self->downstream_flow_ret;
downstream_error:
{
GST_ERROR_OBJECT (self, "Downstream returned %s",
gst_flow_get_name (self->downstream_flow_ret));
if (minfo.data)
gst_buffer_unmap (inbuf, &minfo);
if (inbuf)
gst_buffer_unref (inbuf);
return self->downstream_flow_ret;
}
failed_to_get_input_buffer:
{
GST_ELEMENT_ERROR_FROM_ERROR (self, err);
if (minfo.data)
gst_buffer_unmap (inbuf, &minfo);
if (inbuf)
gst_buffer_unref (inbuf);
return GST_FLOW_ERROR;
}
got_null_input_buffer:
{
GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL),
("Got no input buffer"));
if (minfo.data)
gst_buffer_unmap (inbuf, &minfo);
if (inbuf)
gst_buffer_unref (inbuf);
return GST_FLOW_ERROR;
}
dequeue_error:
{
GST_ELEMENT_ERROR_FROM_ERROR (self, err);
if (minfo.data)
gst_buffer_unmap (inbuf, &minfo);
if (inbuf)
gst_buffer_unref (inbuf);
return GST_FLOW_ERROR;
}
queue_error:
{
GST_AUDIO_DECODER_ERROR_FROM_ERROR (self, err);
if (minfo.data)
gst_buffer_unmap (inbuf, &minfo);
if (inbuf)
gst_buffer_unref (inbuf);
return GST_FLOW_ERROR;
}
flushing:
{
GST_DEBUG_OBJECT (self, "Flushing -- returning FLUSHING");
if (minfo.data)
gst_buffer_unmap (inbuf, &minfo);
if (inbuf)
gst_buffer_unref (inbuf);
return GST_FLOW_FLUSHING;
}
}
static GstFlowReturn
gst_amc_audio_dec_drain (GstAmcAudioDec * self)
{
GstFlowReturn ret;
gint idx;
GError *err = NULL;
GST_DEBUG_OBJECT (self, "Draining codec");
if (!self->started) {
GST_DEBUG_OBJECT (self, "Codec not started yet");
return GST_FLOW_OK;
}
/* Don't send drain buffer twice, this doesn't work */
if (self->drained) {
GST_DEBUG_OBJECT (self, "Codec is drained already");
return GST_FLOW_OK;
}
/* Make sure to release the base class stream lock, otherwise
* _loop() can't call _finish_frame() and we might block forever
* because no input buffers are released */
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
/* Send an EOS buffer to the component and let the base
* class drop the EOS event. We will send it later when
* the EOS buffer arrives on the output port.
* Wait at most 0.5s here. */
idx = gst_amc_codec_dequeue_input_buffer (self->codec, 500000, &err);
GST_AUDIO_DECODER_STREAM_LOCK (self);
if (idx >= 0) {
GstAmcBuffer *buf;
GstAmcBufferInfo buffer_info;
buf = gst_amc_codec_get_input_buffer (self->codec, idx, &err);
if (buf) {
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
g_mutex_lock (&self->drain_lock);
self->draining = TRUE;
memset (&buffer_info, 0, sizeof (buffer_info));
buffer_info.size = 0;
buffer_info.presentation_time_us =
gst_util_uint64_scale (self->last_upstream_ts, 1, GST_USECOND);
buffer_info.flags |= BUFFER_FLAG_END_OF_STREAM;
gst_amc_buffer_set_position_and_limit (buf, NULL, 0, 0);
gst_amc_buffer_free (buf);
buf = NULL;
if (gst_amc_codec_queue_input_buffer (self->codec, idx, &buffer_info,
&err)) {
GST_DEBUG_OBJECT (self, "Waiting until codec is drained");
g_cond_wait (&self->drain_cond, &self->drain_lock);
GST_DEBUG_OBJECT (self, "Drained codec");
ret = GST_FLOW_OK;
} else {
GST_ERROR_OBJECT (self, "Failed to queue input buffer");
if (self->flushing) {
g_clear_error (&err);
ret = GST_FLOW_FLUSHING;
} else {
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
ret = GST_FLOW_ERROR;
}
}
self->drained = TRUE;
self->draining = FALSE;
g_mutex_unlock (&self->drain_lock);
GST_AUDIO_DECODER_STREAM_LOCK (self);
} else {
GST_ERROR_OBJECT (self, "Failed to get buffer for EOS: %d", idx);
if (err)
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
ret = GST_FLOW_ERROR;
}
} else {
GST_ERROR_OBJECT (self, "Failed to acquire buffer for EOS: %d", idx);
if (err)
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
ret = GST_FLOW_ERROR;
}
gst_adapter_flush (self->output_adapter,
gst_adapter_available (self->output_adapter));
return ret;
}
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