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omxaudioenc: Use audio base classes from gst-plugins-base instead of having our own copies

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This commit is contained in:
Sebastian Dröge 2011-11-25 11:31:58 +01:00
parent 9917fbe4c5
commit 217ac7b3be
11 changed files with 96 additions and 4456 deletions
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2
configure.ac
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@ -54,7 +54,7 @@ AC_LIBTOOL_WIN32_DLL
AM_PROG_LIBTOOL AM_PROG_LIBTOOL
dnl *** required versions of GStreamer stuff *** dnl *** required versions of GStreamer stuff ***
GST_REQ=0.10.29 GST_REQ=0.10.35.1
dnl *** autotools stuff **** dnl *** autotools stuff ****

16
omx/Makefile.am
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@ -16,10 +16,7 @@ libgstopenmax_la_SOURCES = \
gstbasevideocodec.c \ gstbasevideocodec.c \
gstbasevideodecoder.c \ gstbasevideodecoder.c \
gstbasevideoencoder.c \ gstbasevideoencoder.c \
gstbasevideoutils.c \ gstbasevideoutils.c
gstbaseaudiodecoder.c \
gstbaseaudioencoder.c \
gstbaseaudioutils.c
noinst_HEADERS = \ noinst_HEADERS = \
gstomx.h \ gstomx.h \
@ -37,10 +34,7 @@ noinst_HEADERS = \
gstbasevideocodec.h \ gstbasevideocodec.h \
gstbasevideodecoder.h \ gstbasevideodecoder.h \
gstbasevideoencoder.h \ gstbasevideoencoder.h \
gstbasevideoutils.h \ gstbasevideoutils.h
gstbaseaudiodecoder.h \
gstbaseaudioencoder.h \
gstbaseaudioutils.h
fixbaseclasses = \ fixbaseclasses = \
-DGstBaseVideoCodec=OMXBaseVideoCodec \ -DGstBaseVideoCodec=OMXBaseVideoCodec \
@ -48,11 +42,7 @@ fixbaseclasses = \
-DGstBaseVideoEncoder=OMXBaseVideoEncoder \ -DGstBaseVideoEncoder=OMXBaseVideoEncoder \
-DGstBaseVideoEncoderClass=OMXBaseVideoEncoderClass \ -DGstBaseVideoEncoderClass=OMXBaseVideoEncoderClass \
-DGstBaseVideoDecoder=OMXBaseVideoDecoder \ -DGstBaseVideoDecoder=OMXBaseVideoDecoder \
-DGstBaseVideoDecoderClass=OMXBaseVideoDecoderClass \ -DGstBaseVideoDecoderClass=OMXBaseVideoDecoderClass
-DGstBaseAudioDecoder=OMXBaseAudioDecoder \
-DGstBaseAudioDecoderClass=OMXBaseAudioDecoderClass \
-DGstBaseAudioEncoder=OMXBaseAudioEncoder \
-DGstBaseAudioEncoderClass=OMXBaseAudioEncoderClass
libgstopenmax_la_CFLAGS = \ libgstopenmax_la_CFLAGS = \
-DGST_USE_UNSTABLE_API=1 \ -DGST_USE_UNSTABLE_API=1 \

1947
omx/gstbaseaudiodecoder.c
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270
omx/gstbaseaudiodecoder.h
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@ -1,270 +0,0 @@
/* GStreamer
* Copyright (C) 2009 Igalia S.L.
* Author: Iago Toral Quiroga <itoral@igalia.com>
* Copyright (C) 2011 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>.
* Copyright (C) 2011 Nokia Corporation. All rights reserved.
* Contact: Stefan Kost <stefan.kost@nokia.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* 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
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifndef _GST_BASE_AUDIO_DECODER_H_
#define _GST_BASE_AUDIO_DECODER_H_
#ifndef GST_USE_UNSTABLE_API
#warning "GstBaseAudioDecoder is unstable API and may change in future."
#warning "You can define GST_USE_UNSTABLE_API to avoid this warning."
#endif
#include <gst/gst.h>
#include <gst/base/gstadapter.h>
#include "gstbaseaudioutils.h"
G_BEGIN_DECLS
#define GST_TYPE_BASE_AUDIO_DECODER \
(gst_base_audio_decoder_get_type())
#define GST_BASE_AUDIO_DECODER(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_BASE_AUDIO_DECODER,GstBaseAudioDecoder))
#define GST_BASE_AUDIO_DECODER_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_BASE_AUDIO_DECODER,GstBaseAudioDecoderClass))
#define GST_BASE_AUDIO_DECODER_GET_CLASS(obj) \
(G_TYPE_INSTANCE_GET_CLASS((obj),GST_TYPE_BASE_AUDIO_DECODER,GstBaseAudioDecoderClass))
#define GST_IS_BASE_AUDIO_DECODER(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_BASE_AUDIO_DECODER))
#define GST_IS_BASE_AUDIO_DECODER_CLASS(obj) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_BASE_AUDIO_DECODER))
/**
* GST_BASE_AUDIO_DECODER_SINK_NAME:
*
* The name of the templates for the sink pad.
*/
#define GST_BASE_AUDIO_DECODER_SINK_NAME "sink"
/**
* GST_BASE_AUDIO_DECODER_SRC_NAME:
*
* The name of the templates for the source pad.
*/
#define GST_BASE_AUDIO_DECODER_SRC_NAME "src"
/**
* GST_BASE_AUDIO_DECODER_SRC_PAD:
* @obj: base audio codec instance
*
* Gives the pointer to the source #GstPad object of the element.
*/
#define GST_BASE_AUDIO_DECODER_SRC_PAD(obj) (((GstBaseAudioDecoder *) (obj))->srcpad)
/**
* GST_BASE_AUDIO_DECODER_SINK_PAD:
* @obj: base audio codec instance
*
* Gives the pointer to the sink #GstPad object of the element.
*/
#define GST_BASE_AUDIO_DECODER_SINK_PAD(obj) (((GstBaseAudioDecoder *) (obj))->sinkpad)
#define GST_BASE_AUDIO_DECODER_STREAM_LOCK(dec) g_static_rec_mutex_lock (&GST_BASE_AUDIO_DECODER (dec)->stream_lock)
#define GST_BASE_AUDIO_DECODER_STREAM_UNLOCK(dec) g_static_rec_mutex_unlock (&GST_BASE_AUDIO_DECODER (dec)->stream_lock)
typedef struct _GstBaseAudioDecoder GstBaseAudioDecoder;
typedef struct _GstBaseAudioDecoderClass GstBaseAudioDecoderClass;
typedef struct _GstBaseAudioDecoderPrivate GstBaseAudioDecoderPrivate;
typedef struct _GstBaseAudioDecoderContext GstBaseAudioDecoderContext;
/* do not use this one, use macro below */
GstFlowReturn _gst_base_audio_decoder_error (GstBaseAudioDecoder *dec, gint weight,
GQuark domain, gint code,
gchar *txt, gchar *debug,
const gchar *file, const gchar *function,
gint line);
/**
* GST_BASE_AUDIO_DECODER_ERROR:
* @el: the base audio decoder element that generates the error
* @weight: element defined weight of the error, added to error count
* @domain: like CORE, LIBRARY, RESOURCE or STREAM (see #gstreamer-GstGError)
* @code: error code defined for that domain (see #gstreamer-GstGError)
* @text: the message to display (format string and args enclosed in
* parentheses)
* @debug: debugging information for the message (format string and args
* enclosed in parentheses)
* @ret: variable to receive return value
*
* Utility function that audio decoder elements can use in case they encountered
* a data processing error that may be fatal for the current "data unit" but
* need not prevent subsequent decoding. Such errors are counted and if there
* are too many, as configured in the context's max_errors, the pipeline will
* post an error message and the application will be requested to stop further
* media processing. Otherwise, it is considered a "glitch" and only a warning
* is logged. In either case, @ret is set to the proper value to
* return to upstream/caller (indicating either GST_FLOW_ERROR or GST_FLOW_OK).
*/
#define GST_BASE_AUDIO_DECODER_ERROR(el, w, domain, code, text, debug, ret) \
G_STMT_START { \
gchar *__txt = _gst_element_error_printf text; \
gchar *__dbg = _gst_element_error_printf debug; \
GstBaseAudioDecoder *dec = GST_BASE_AUDIO_DECODER (el); \
ret = _gst_base_audio_decoder_error (dec, w, GST_ ## domain ## _ERROR, \
GST_ ## domain ## _ERROR_ ## code, __txt, __dbg, __FILE__, \
GST_FUNCTION, __LINE__); \
} G_STMT_END
/**
* GstBaseAudioDecoderContext:
* @state: a #GstAudioState describing input audio format
* @eos: no (immediate) subsequent data in stream
* @sync: stream parsing in sync
* @delay: number of frames pending decoding (typically at least 1 for current)
* @do_plc: whether subclass is prepared to handle (packet) loss concealment
* @min_latency: min latency of element
* @max_latency: max latency of element
* @lookahead: decoder lookahead (in units of input rate samples)
*
* Transparent #GstBaseAudioEncoderContext data structure.
*/
struct _GstBaseAudioDecoderContext {
/* input */
/* (output) audio format */
GstAudioState state;
/* parsing state */
gboolean eos;
gboolean sync;
/* misc */
gint delay;
/* output */
gboolean do_plc;
gboolean do_byte_time;
gint max_errors;
/* MT-protected (with LOCK) */
GstClockTime min_latency;
GstClockTime max_latency;
};
/**
* GstBaseAudioDecoder:
*
* The opaque #GstBaseAudioDecoder data structure.
*/
struct _GstBaseAudioDecoder
{
GstElement element;
/*< protected >*/
/* source and sink pads */
GstPad *sinkpad;
GstPad *srcpad;
/* protects all data processing, i.e. is locked
* in the chain function, finish_frame and when
* processing serialized events */
GStaticRecMutex stream_lock;
/* MT-protected (with STREAM_LOCK) */
GstSegment segment;
GstBaseAudioDecoderContext *ctx;
/* properties */
GstClockTime latency;
GstClockTime tolerance;
gboolean plc;
/*< private >*/
GstBaseAudioDecoderPrivate *priv;
gpointer _gst_reserved[GST_PADDING_LARGE];
};
/**
* GstBaseAudioDecoderClass:
* @start: Optional.
* Called when the element starts processing.
* Allows opening external resources.
* @stop: Optional.
* Called when the element stops processing.
* Allows closing external resources.
* @set_format: Notifies subclass of incoming data format (caps).
* @parse: Optional.
* Allows chopping incoming data into manageable units (frames)
* for subsequent decoding. This division is at subclass
* discretion and may or may not correspond to 1 (or more)
* frames as defined by audio format.
* @handle_frame: Provides input data (or NULL to clear any remaining data)
* to subclass. Input data ref management is performed by
* base class, subclass should not care or intervene.
* @flush: Optional.
* Instructs subclass to clear any codec caches and discard
* any pending samples and not yet returned encoded data.
* @hard indicates whether a FLUSH is being processed,
* or otherwise a DISCONT (or conceptually similar).
* @event: Optional.
* Event handler on the sink pad. This function should return
* TRUE if the event was handled and should be discarded
* (i.e. not unref'ed).
* @pre_push: Optional.
* Called just prior to pushing (encoded data) buffer downstream.
* Subclass has full discretionary access to buffer,
* and a not OK flow return will abort downstream pushing.
*
* Subclasses can override any of the available virtual methods or not, as
* needed. At minimum @handle_frame (and likely @set_format) needs to be
* overridden.
*/
struct _GstBaseAudioDecoderClass
{
GstElementClass parent_class;
/*< public >*/
/* virtual methods for subclasses */
gboolean (*start) (GstBaseAudioDecoder *dec);
gboolean (*stop) (GstBaseAudioDecoder *dec);
gboolean (*set_format) (GstBaseAudioDecoder *dec,
GstCaps *caps);
GstFlowReturn (*parse) (GstBaseAudioDecoder *dec,
GstAdapter *adapter,
gint *offset, gint *length);
GstFlowReturn (*handle_frame) (GstBaseAudioDecoder *dec,
GstBuffer *buffer);
void (*flush) (GstBaseAudioDecoder *dec, gboolean hard);
GstFlowReturn (*pre_push) (GstBaseAudioDecoder *dec,
GstBuffer **buffer);
gboolean (*event) (GstBaseAudioDecoder *dec,
GstEvent *event);
/*< private >*/
gpointer _gst_reserved[GST_PADDING_LARGE];
};
GstFlowReturn gst_base_audio_decoder_finish_frame (GstBaseAudioDecoder * dec,
GstBuffer * buf, gint frames);
GType gst_base_audio_decoder_get_type (void);
G_END_DECLS
#endif

1507
omx/gstbaseaudioencoder.c
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234
omx/gstbaseaudioencoder.h
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@ -1,234 +0,0 @@
/* GStreamer
* Copyright (C) 2011 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>.
* Copyright (C) 2011 Nokia Corporation. All rights reserved.
* Contact: Stefan Kost <stefan.kost@nokia.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* 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
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifndef __GST_BASE_AUDIO_ENCODER_H__
#define __GST_BASE_AUDIO_ENCODER_H__
#ifndef GST_USE_UNSTABLE_API
#warning "GstBaseAudioEncoder is unstable API and may change in future."
#warning "You can define GST_USE_UNSTABLE_API to avoid this warning."
#endif
#include <gst/gst.h>
#include "gstbaseaudioutils.h"
G_BEGIN_DECLS
#define GST_TYPE_BASE_AUDIO_ENCODER (gst_base_audio_encoder_get_type())
#define GST_BASE_AUDIO_ENCODER(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_BASE_AUDIO_ENCODER,GstBaseAudioEncoder))
#define GST_BASE_AUDIO_ENCODER_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_BASE_AUDIO_ENCODER,GstBaseAudioEncoderClass))
#define GST_BASE_AUDIO_ENCODER_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj),GST_TYPE_BASE_AUDIO_ENCODER,GstBaseAudioEncoderClass))
#define GST_IS_BASE_AUDIO_ENCODER(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_BASE_AUDIO_ENCODER))
#define GST_IS_BASE_AUDIO_ENCODER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_BASE_AUDIO_ENCODER))
#define GST_BASE_AUDIO_ENCODER_CAST(obj) ((GstBaseAudioEncoder *)(obj))
/**
* GST_BASE_AUDIO_ENCODER_SINK_NAME:
*
* the name of the templates for the sink pad
*/
#define GST_BASE_AUDIO_ENCODER_SINK_NAME "sink"
/**
* GST_BASE_AUDIO_ENCODER_SRC_NAME:
*
* the name of the templates for the source pad
*/
#define GST_BASE_AUDIO_ENCODER_SRC_NAME "src"
/**
* GST_BASE_AUDIO_ENCODER_SRC_PAD:
* @obj: base parse instance
*
* Gives the pointer to the source #GstPad object of the element.
*
* Since: 0.10.x
*/
#define GST_BASE_AUDIO_ENCODER_SRC_PAD(obj) (GST_BASE_AUDIO_ENCODER_CAST (obj)->srcpad)
/**
* GST_BASE_AUDIO_ENCODER_SINK_PAD:
* @obj: base parse instance
*
* Gives the pointer to the sink #GstPad object of the element.
*
* Since: 0.10.x
*/
#define GST_BASE_AUDIO_ENCODER_SINK_PAD(obj) (GST_BASE_AUDIO_ENCODER_CAST (obj)->sinkpad)
/**
* GST_BASE_AUDIO_ENCODER_SEGMENT:
* @obj: base parse instance
*
* Gives the segment of the element.
*
* Since: 0.10.x
*/
#define GST_BASE_AUDIO_ENCODER_SEGMENT(obj) (GST_BASE_AUDIO_ENCODER_CAST (obj)->segment)
#define GST_BASE_AUDIO_ENCODER_STREAM_LOCK(enc) g_static_rec_mutex_lock (&GST_BASE_AUDIO_ENCODER (enc)->stream_lock)
#define GST_BASE_AUDIO_ENCODER_STREAM_UNLOCK(enc) g_static_rec_mutex_unlock (&GST_BASE_AUDIO_ENCODER (enc)->stream_lock)
typedef struct _GstBaseAudioEncoder GstBaseAudioEncoder;
typedef struct _GstBaseAudioEncoderClass GstBaseAudioEncoderClass;
typedef struct _GstBaseAudioEncoderPrivate GstBaseAudioEncoderPrivate;
typedef struct _GstBaseAudioEncoderContext GstBaseAudioEncoderContext;
/**
* GstBaseAudioEncoderContext:
* @state: a #GstAudioState describing input audio format
* @frame_samples_min: number of samples (per channel) subclass needs to be handed
* at least, or will be handed all available if 0.
* @frame_samples_max: number of samples (per channel) subclass needs to be handed
* at most, or will be handed all available if 0.
* @frame_max: max number of frames of size @frame_samples accepted at once
* (assumed minimally 1). Requires @frame_samples_min and @frame_samples_max
* to be the equal.
* @min_latency: min latency of element
* @max_latency: max latency of element
* @lookahead: encoder lookahead (in units of input rate samples)
*
* Transparent #GstBaseAudioEncoderContext data structure.
*/
struct _GstBaseAudioEncoderContext {
/* input */
GstAudioState state;
/* output */
gint frame_samples_min, frame_samples_max;
gint frame_max;
gint lookahead;
/* MT-protected (with LOCK) */
GstClockTime min_latency;
GstClockTime max_latency;
};
/**
* GstBaseAudioEncoder:
* @element: the parent element.
*
* The opaque #GstBaseAudioEncoder data structure.
*/
struct _GstBaseAudioEncoder {
GstElement element;
/*< protected >*/
/* source and sink pads */
GstPad *sinkpad;
GstPad *srcpad;
/* protects all data processing, i.e. is locked
* in the chain function, finish_frame and when
* processing serialized events */
GStaticRecMutex stream_lock;
/* MT-protected (with STREAM_LOCK) */
GstSegment segment;
GstBaseAudioEncoderContext *ctx;
/* properties */
gint64 tolerance;
gboolean perfect_ts;
gboolean hard_resync;
gboolean granule;
/*< private >*/
GstBaseAudioEncoderPrivate *priv;
gpointer _gst_reserved[GST_PADDING_LARGE];
};
/**
* GstBaseAudioEncoderClass:
* @start: Optional.
* Called when the element starts processing.
* Allows opening external resources.
* @stop: Optional.
* Called when the element stops processing.
* Allows closing external resources.
* @set_format: Notifies subclass of incoming data format.
* GstBaseAudioEncoderContext fields have already been
* set according to provided caps.
* @handle_frame: Provides input samples (or NULL to clear any remaining data)
* according to directions as provided by subclass in the
* #GstBaseAudioEncoderContext. Input data ref management
* is performed by base class, subclass should not care or
* intervene.
* @flush: Optional.
* Instructs subclass to clear any codec caches and discard
* any pending samples and not yet returned encoded data.
* @event: Optional.
* Event handler on the sink pad. This function should return
* TRUE if the event was handled and should be discarded
* (i.e. not unref'ed).
* @pre_push: Optional.
* Called just prior to pushing (encoded data) buffer downstream.
* Subclass has full discretionary access to buffer,
* and a not OK flow return will abort downstream pushing.
* @getcaps: Optional.
* Allows for a custom sink getcaps implementation (e.g.
* for multichannel input specification). If not implemented,
* default returns gst_base_audio_encoder_proxy_getcaps
* applied to sink template caps.
*
* Subclasses can override any of the available virtual methods or not, as
* needed. At minimum @set_format and @handle_frame needs to be overridden.
*/
struct _GstBaseAudioEncoderClass {
GstElementClass parent_class;
/*< public >*/
/* virtual methods for subclasses */
gboolean (*start) (GstBaseAudioEncoder *enc);
gboolean (*stop) (GstBaseAudioEncoder *enc);
gboolean (*set_format) (GstBaseAudioEncoder *enc,
GstAudioState *state);
GstFlowReturn (*handle_frame) (GstBaseAudioEncoder *enc,
GstBuffer *buffer);
void (*flush) (GstBaseAudioEncoder *enc);
GstFlowReturn (*pre_push) (GstBaseAudioEncoder *enc,
GstBuffer **buffer);
gboolean (*event) (GstBaseAudioEncoder *enc,
GstEvent *event);
GstCaps * (*getcaps) (GstBaseAudioEncoder *enc);
/*< private >*/
gpointer _gst_reserved[GST_PADDING_LARGE];
};
GType gst_base_audio_encoder_get_type (void);
GstFlowReturn gst_base_audio_encoder_finish_frame (GstBaseAudioEncoder * enc,
GstBuffer *buffer, gint samples);
GstCaps * gst_base_audio_encoder_proxy_getcaps (GstBaseAudioEncoder * enc,
GstCaps * caps);
G_END_DECLS
#endif /* __GST_BASE_AUDIO_ENCODER_H__ */

315
omx/gstbaseaudioutils.c
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@ -1,315 +0,0 @@
/* GStreamer
* Copyright (C) 2011 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>.
* Copyright (C) 2011 Nokia Corporation. All rights reserved.
* Contact: Stefan Kost <stefan.kost@nokia.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* 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
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#include "gstbaseaudioutils.h"
#include <gst/gst.h>
#include <gst/audio/multichannel.h>
#define CHECK_VALUE(var, val) \
G_STMT_START { \
if (!res) \
goto fail; \
if (var != val) \
changed = TRUE; \
var = val; \
} G_STMT_END
/**
* gst_base_audio_parse_caps:
* @caps: a #GstCaps
* @state: a #GstAudioState
* @changed: whether @caps introduced a change in current @state
*
* Parses audio format as represented by @caps into a more concise form
* as represented by @state, while checking if for changes to currently
* defined audio format.
*
* Returns: TRUE if parsing succeeded, otherwise FALSE
*/
gboolean
gst_base_audio_parse_caps (GstCaps * caps, GstAudioState * state,
gboolean * _changed)
{
gboolean res = TRUE, changed = FALSE;
GstStructure *s;
gboolean vb;
gint vi;
g_return_val_if_fail (caps != NULL, FALSE);
g_return_val_if_fail (gst_caps_is_fixed (caps), FALSE);
s = gst_caps_get_structure (caps, 0);
if (gst_structure_has_name (s, "audio/x-raw-int"))
state->is_int = TRUE;
else if (gst_structure_has_name (s, "audio/x-raw-float"))
state->is_int = FALSE;
else
goto fail;
res = gst_structure_get_int (s, "rate", &vi);
CHECK_VALUE (state->rate, vi);
res &= gst_structure_get_int (s, "channels", &vi);
CHECK_VALUE (state->channels, vi);
res &= gst_structure_get_int (s, "width", &vi);
CHECK_VALUE (state->width, vi);
res &= (!state->is_int || gst_structure_get_int (s, "depth", &vi));
CHECK_VALUE (state->depth, vi);
res &= gst_structure_get_int (s, "endianness", &vi);
CHECK_VALUE (state->endian, vi);
res &= (!state->is_int || gst_structure_get_boolean (s, "signed", &vb));
CHECK_VALUE (state->sign, vb);
state->bpf = (state->width / 8) * state->channels;
GST_LOG ("bpf: %d", state->bpf);
if (!state->bpf)
goto fail;
g_free (state->channel_pos);
state->channel_pos = gst_audio_get_channel_positions (s);
if (_changed)
*_changed = changed;
return res;
/* ERRORS */
fail:
{
/* there should not be caps out there that fail parsing ... */
GST_WARNING ("failed to parse caps %" GST_PTR_FORMAT, caps);
return res;
}
}
/**
* gst_base_audio_add_streamheader:
* @caps: a #GstCaps
* @buf: header buffers
*
* Adds given buffers to an array of buffers set as streamheader field
* on the given @caps. List of buffer arguments must be NULL-terminated.
*
* Returns: input caps with a streamheader field added, or NULL if some error
*/
GstCaps *
gst_base_audio_add_streamheader (GstCaps * caps, GstBuffer * buf, ...)
{
GstStructure *structure = NULL;
va_list va;
GValue array = { 0 };
GValue value = { 0 };
g_return_val_if_fail (caps != NULL, NULL);
g_return_val_if_fail (gst_caps_is_fixed (caps), NULL);
caps = gst_caps_make_writable (caps);
structure = gst_caps_get_structure (caps, 0);
g_value_init (&array, GST_TYPE_ARRAY);
va_start (va, buf);
/* put buffers in a fixed list */
while (buf) {
g_assert (gst_buffer_is_metadata_writable (buf));
/* mark buffer */
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_IN_CAPS);
g_value_init (&value, GST_TYPE_BUFFER);
buf = gst_buffer_copy (buf);
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_IN_CAPS);
gst_value_set_buffer (&value, buf);
gst_buffer_unref (buf);
gst_value_array_append_value (&array, &value);
g_value_unset (&value);
buf = va_arg (va, GstBuffer *);
}
gst_structure_set_value (structure, "streamheader", &array);
g_value_unset (&array);
return caps;
}
/**
* gst_base_audio_encoded_audio_convert:
* @fmt: audio format of the encoded audio
* @bytes: number of encoded bytes
* @samples: number of encoded samples
* @src_format: source format
* @src_value: source value
* @dest_format: destination format
* @dest_value: destination format
*
* Helper function to convert @src_value in @src_format to @dest_value in
* @dest_format for encoded audio data. Conversion is possible between
* BYTE and TIME format by using estimated bitrate based on
* @samples and @bytes (and @fmt).
*/
gboolean
gst_base_audio_encoded_audio_convert (GstAudioState * fmt,
gint64 bytes, gint64 samples, GstFormat src_format,
gint64 src_value, GstFormat * dest_format, gint64 * dest_value)
{
gboolean res = FALSE;
g_return_val_if_fail (dest_format != NULL, FALSE);
g_return_val_if_fail (dest_value != NULL, FALSE);
if (G_UNLIKELY (src_format == *dest_format || src_value == 0 ||
src_value == -1)) {
if (dest_value)
*dest_value = src_value;
return TRUE;
}
if (samples == 0 || bytes == 0 || fmt->rate == 0) {
GST_DEBUG ("not enough metadata yet to convert");
goto exit;
}
bytes *= fmt->rate;
switch (src_format) {
case GST_FORMAT_BYTES:
switch (*dest_format) {
case GST_FORMAT_TIME:
*dest_value = gst_util_uint64_scale (src_value,
GST_SECOND * samples, bytes);
res = TRUE;
break;
default:
res = FALSE;
}
break;
case GST_FORMAT_TIME:
switch (*dest_format) {
case GST_FORMAT_BYTES:
*dest_value = gst_util_uint64_scale (src_value, bytes,
samples * GST_SECOND);
res = TRUE;
break;
default:
res = FALSE;
}
break;
default:
res = FALSE;
}
exit:
return res;
}
/**
* gst_base_audio_raw_audio_convert:
* @fmt: audio format of the encoded audio
* @src_format: source format
* @src_value: source value
* @dest_format: destination format
* @dest_value: destination format
*
* Helper function to convert @src_value in @src_format to @dest_value in
* @dest_format for encoded audio data. Conversion is possible between
* BYTE, DEFAULT and TIME format based on audio characteristics provided
* by @fmt.
*/
gboolean
gst_base_audio_raw_audio_convert (GstAudioState * fmt, GstFormat src_format,
gint64 src_value, GstFormat * dest_format, gint64 * dest_value)
{
gboolean res = FALSE;
guint scale = 1;
gint bytes_per_sample, rate, byterate;
g_return_val_if_fail (dest_format != NULL, FALSE);
g_return_val_if_fail (dest_value != NULL, FALSE);
if (G_UNLIKELY (src_format == *dest_format || src_value == 0 ||
src_value == -1)) {
if (dest_value)
*dest_value = src_value;
return TRUE;
}
bytes_per_sample = fmt->bpf;
rate = fmt->rate;
byterate = bytes_per_sample * rate;
if (G_UNLIKELY (bytes_per_sample == 0 || rate == 0)) {
GST_DEBUG ("not enough metadata yet to convert");
goto exit;
}
switch (src_format) {
case GST_FORMAT_BYTES:
switch (*dest_format) {
case GST_FORMAT_DEFAULT:
*dest_value = src_value / bytes_per_sample;
res = TRUE;
break;
case GST_FORMAT_TIME:
*dest_value =
gst_util_uint64_scale_int (src_value, GST_SECOND, byterate);
res = TRUE;
break;
default:
res = FALSE;
}
break;
case GST_FORMAT_DEFAULT:
switch (*dest_format) {
case GST_FORMAT_BYTES:
*dest_value = src_value * bytes_per_sample;
res = TRUE;
break;
case GST_FORMAT_TIME:
*dest_value = gst_util_uint64_scale_int (src_value, GST_SECOND, rate);
res = TRUE;
break;
default:
res = FALSE;
}
break;
case GST_FORMAT_TIME:
switch (*dest_format) {
case GST_FORMAT_BYTES:
scale = bytes_per_sample;
/* fallthrough */
case GST_FORMAT_DEFAULT:
*dest_value = gst_util_uint64_scale_int (src_value,
scale * rate, GST_SECOND);
res = TRUE;
break;
default:
res = FALSE;
}
break;
default:
res = FALSE;
}
exit:
return res;
}

74
omx/gstbaseaudioutils.h
View file

@ -1,74 +0,0 @@
/* GStreamer
* Copyright (C) 2011 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>.
* Copyright (C) 2011 Nokia Corporation. All rights reserved.
* Contact: Stefan Kost <stefan.kost@nokia.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* 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
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifndef _GST_BASE_AUDIO_UTILS_H_
#define _GST_BASE_AUDIO_UTILS_H_
#ifndef GST_USE_UNSTABLE_API
#warning "Base audio utils provide unstable API and may change in future."
#warning "You can define GST_USE_UNSTABLE_API to avoid this warning."
#endif
#include <gst/gst.h>
#include <gst/audio/multichannel.h>
G_BEGIN_DECLS
/**
* GstAudioState:
* @is_int: whether sample data is int or float
* @rate: rate of sample data
* @channels: number of channels in sample data
* @width: width (in bits) of sample data
* @depth: used bits in sample data (if integer)
* @sign: sign of sample data (if integer)
* @endian: endianness of sample data
* @bpf: bytes per audio frame
*/
typedef struct _GstAudioState {
gboolean is_int;
gint rate;
gint channels;
gint width;
gint depth;
gboolean sign;
gint endian;
GstAudioChannelPosition *channel_pos;
gint bpf;
} GstAudioState;
gboolean gst_base_audio_parse_caps (GstCaps * caps,
GstAudioState * state, gboolean * changed);
GstCaps *gst_base_audio_add_streamheader (GstCaps * caps, GstBuffer * buf, ...);
gboolean gst_base_audio_encoded_audio_convert (GstAudioState * fmt,
gint64 bytes, gint64 samples, GstFormat src_format,
gint64 src_value, GstFormat * dest_format, gint64 * dest_value);
gboolean gst_base_audio_raw_audio_convert (GstAudioState * fmt, GstFormat src_format,
gint64 src_value, GstFormat * dest_format, gint64 * dest_value);
G_END_DECLS
#endif

16
omx/gstomxaacenc.c
View file

@ -36,11 +36,11 @@ static void gst_omx_aac_enc_set_property (GObject * object, guint prop_id,
static void gst_omx_aac_enc_get_property (GObject * object, guint prop_id, static void gst_omx_aac_enc_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec); GValue * value, GParamSpec * pspec);
static gboolean gst_omx_aac_enc_set_format (GstOMXAudioEnc * enc, static gboolean gst_omx_aac_enc_set_format (GstOMXAudioEnc * enc,
GstOMXPort * port, GstAudioState * state); GstOMXPort * port, GstAudioInfo * info);
static GstCaps *gst_omx_aac_enc_get_caps (GstOMXAudioEnc * enc, static GstCaps *gst_omx_aac_enc_get_caps (GstOMXAudioEnc * enc,
GstOMXPort * port, GstAudioState * state); GstOMXPort * port, GstAudioInfo * info);
static guint gst_omx_aac_enc_get_num_samples (GstOMXAudioEnc * enc, static guint gst_omx_aac_enc_get_num_samples (GstOMXAudioEnc * enc,
GstOMXPort * port, GstAudioState * state, GstOMXBuffer * buf); GstOMXPort * port, GstAudioInfo * info, GstOMXBuffer * buf);
enum enum
{ {
@ -228,7 +228,7 @@ gst_omx_aac_enc_get_property (GObject * object, guint prop_id, GValue * value,
static gboolean static gboolean
gst_omx_aac_enc_set_format (GstOMXAudioEnc * enc, GstOMXPort * port, gst_omx_aac_enc_set_format (GstOMXAudioEnc * enc, GstOMXPort * port,
GstAudioState * state) GstAudioInfo * info)
{ {
GstOMXAACEnc *self = GST_OMX_AAC_ENC (enc); GstOMXAACEnc *self = GST_OMX_AAC_ENC (enc);
OMX_AUDIO_PARAM_AACPROFILETYPE aac_profile; OMX_AUDIO_PARAM_AACPROFILETYPE aac_profile;
@ -250,7 +250,7 @@ gst_omx_aac_enc_set_format (GstOMXAudioEnc * enc, GstOMXPort * port,
return FALSE; return FALSE;
} }
peercaps = gst_pad_peer_get_caps (GST_BASE_AUDIO_ENCODER_SRC_PAD (self)); peercaps = gst_pad_peer_get_caps (GST_AUDIO_ENCODER_SRC_PAD (self));
if (peercaps) { if (peercaps) {
GstCaps *intersection; GstCaps *intersection;
GstStructure *s; GstStructure *s;
@ -259,7 +259,7 @@ gst_omx_aac_enc_set_format (GstOMXAudioEnc * enc, GstOMXPort * port,
intersection = intersection =
gst_caps_intersect (peercaps, gst_caps_intersect (peercaps,
gst_pad_get_pad_template_caps (GST_BASE_AUDIO_ENCODER_SRC_PAD (self))); gst_pad_get_pad_template_caps (GST_AUDIO_ENCODER_SRC_PAD (self)));
gst_caps_unref (peercaps); gst_caps_unref (peercaps);
if (gst_caps_is_empty (intersection)) { if (gst_caps_is_empty (intersection)) {
gst_caps_unref (intersection); gst_caps_unref (intersection);
@ -340,7 +340,7 @@ gst_omx_aac_enc_set_format (GstOMXAudioEnc * enc, GstOMXPort * port,
static GstCaps * static GstCaps *
gst_omx_aac_enc_get_caps (GstOMXAudioEnc * enc, GstOMXPort * port, gst_omx_aac_enc_get_caps (GstOMXAudioEnc * enc, GstOMXPort * port,
GstAudioState * state) GstAudioInfo * info)
{ {
GstCaps *caps; GstCaps *caps;
OMX_ERRORTYPE err; OMX_ERRORTYPE err;
@ -437,7 +437,7 @@ gst_omx_aac_enc_get_caps (GstOMXAudioEnc * enc, GstOMXPort * port,
static guint static guint
gst_omx_aac_enc_get_num_samples (GstOMXAudioEnc * enc, GstOMXPort * port, gst_omx_aac_enc_get_num_samples (GstOMXAudioEnc * enc, GstOMXPort * port,
GstAudioState * state, GstOMXBuffer * buf) GstAudioInfo * info, GstOMXBuffer * buf)
{ {
/* FIXME: Depends on the profile at least */ /* FIXME: Depends on the profile at least */
return 1024; return 1024;

159
omx/gstomxaudioenc.c
View file

@ -37,15 +37,15 @@ static GstStateChangeReturn
gst_omx_audio_enc_change_state (GstElement * element, gst_omx_audio_enc_change_state (GstElement * element,
GstStateChange transition); GstStateChange transition);
static gboolean gst_omx_audio_enc_start (GstBaseAudioEncoder * encoder); static gboolean gst_omx_audio_enc_start (GstAudioEncoder * encoder);
static gboolean gst_omx_audio_enc_stop (GstBaseAudioEncoder * encoder); static gboolean gst_omx_audio_enc_stop (GstAudioEncoder * encoder);
static gboolean gst_omx_audio_enc_set_format (GstBaseAudioEncoder * encoder, static gboolean gst_omx_audio_enc_set_format (GstAudioEncoder * encoder,
GstAudioState * state); GstAudioInfo * info);
static gboolean gst_omx_audio_enc_event (GstBaseAudioEncoder * encoder, static gboolean gst_omx_audio_enc_event (GstAudioEncoder * encoder,
GstEvent * event); GstEvent * event);
static GstFlowReturn gst_omx_audio_enc_handle_frame (GstBaseAudioEncoder * static GstFlowReturn gst_omx_audio_enc_handle_frame (GstAudioEncoder *
encoder, GstBuffer * buffer); encoder, GstBuffer * buffer);
static void gst_omx_audio_enc_flush (GstBaseAudioEncoder * encoder); static void gst_omx_audio_enc_flush (GstAudioEncoder * encoder);
static GstFlowReturn gst_omx_audio_enc_drain (GstOMXAudioEnc * self); static GstFlowReturn gst_omx_audio_enc_drain (GstOMXAudioEnc * self);
@ -60,8 +60,8 @@ enum
GST_DEBUG_CATEGORY_INIT (gst_omx_audio_enc_debug_category, "omxaudioenc", 0, \ GST_DEBUG_CATEGORY_INIT (gst_omx_audio_enc_debug_category, "omxaudioenc", 0, \
"debug category for gst-omx audio encoder base class"); "debug category for gst-omx audio encoder base class");
GST_BOILERPLATE_FULL (GstOMXAudioEnc, gst_omx_audio_enc, GstBaseAudioEncoder, GST_BOILERPLATE_FULL (GstOMXAudioEnc, gst_omx_audio_enc, GstAudioEncoder,
GST_TYPE_BASE_AUDIO_ENCODER, DEBUG_INIT); GST_TYPE_AUDIO_ENCODER, DEBUG_INIT);
static void static void
gst_omx_audio_enc_base_init (gpointer g_class) gst_omx_audio_enc_base_init (gpointer g_class)
@ -203,22 +203,21 @@ gst_omx_audio_enc_class_init (GstOMXAudioEncClass * klass)
{ {
GObjectClass *gobject_class = G_OBJECT_CLASS (klass); GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass); GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstBaseAudioEncoderClass *base_audio_encoder_class = GstAudioEncoderClass *audio_encoder_class = GST_AUDIO_ENCODER_CLASS (klass);
GST_BASE_AUDIO_ENCODER_CLASS (klass);
gobject_class->finalize = gst_omx_audio_enc_finalize; gobject_class->finalize = gst_omx_audio_enc_finalize;
element_class->change_state = element_class->change_state =
GST_DEBUG_FUNCPTR (gst_omx_audio_enc_change_state); GST_DEBUG_FUNCPTR (gst_omx_audio_enc_change_state);
base_audio_encoder_class->start = GST_DEBUG_FUNCPTR (gst_omx_audio_enc_start); audio_encoder_class->start = GST_DEBUG_FUNCPTR (gst_omx_audio_enc_start);
base_audio_encoder_class->stop = GST_DEBUG_FUNCPTR (gst_omx_audio_enc_stop); audio_encoder_class->stop = GST_DEBUG_FUNCPTR (gst_omx_audio_enc_stop);
base_audio_encoder_class->flush = GST_DEBUG_FUNCPTR (gst_omx_audio_enc_flush); audio_encoder_class->flush = GST_DEBUG_FUNCPTR (gst_omx_audio_enc_flush);
base_audio_encoder_class->set_format = audio_encoder_class->set_format =
GST_DEBUG_FUNCPTR (gst_omx_audio_enc_set_format); GST_DEBUG_FUNCPTR (gst_omx_audio_enc_set_format);
base_audio_encoder_class->handle_frame = audio_encoder_class->handle_frame =
GST_DEBUG_FUNCPTR (gst_omx_audio_enc_handle_frame); GST_DEBUG_FUNCPTR (gst_omx_audio_enc_handle_frame);
base_audio_encoder_class->event = GST_DEBUG_FUNCPTR (gst_omx_audio_enc_event); audio_encoder_class->event = GST_DEBUG_FUNCPTR (gst_omx_audio_enc_event);
klass->default_sink_template_caps = "audio/x-raw-int, " klass->default_sink_template_caps = "audio/x-raw-int, "
"rate = (int) [ 1, MAX ], " "rate = (int) [ 1, MAX ], "
@ -432,31 +431,32 @@ gst_omx_audio_enc_loop (GstOMXAudioEnc * self)
return; return;
} }
if (!GST_PAD_CAPS (GST_BASE_AUDIO_ENCODER_SRC_PAD (self)) if (!GST_PAD_CAPS (GST_AUDIO_ENCODER_SRC_PAD (self))
|| acq_return == GST_OMX_ACQUIRE_BUFFER_RECONFIGURED) { || acq_return == GST_OMX_ACQUIRE_BUFFER_RECONFIGURED) {
GstAudioState *state = &GST_BASE_AUDIO_ENCODER (self)->ctx->state; GstAudioInfo *info =
gst_audio_encoder_get_audio_info (GST_AUDIO_ENCODER (self));
GstCaps *caps; GstCaps *caps;
GST_DEBUG_OBJECT (self, "Port settings have changed, updating caps"); GST_DEBUG_OBJECT (self, "Port settings have changed, updating caps");
GST_BASE_AUDIO_ENCODER_STREAM_LOCK (self); GST_AUDIO_ENCODER_STREAM_LOCK (self);
caps = klass->get_caps (self, self->out_port, state); caps = klass->get_caps (self, self->out_port, info);
if (!caps) { if (!caps) {
if (buf) if (buf)
gst_omx_port_release_buffer (self->out_port, buf); gst_omx_port_release_buffer (self->out_port, buf);
GST_BASE_AUDIO_ENCODER_STREAM_UNLOCK (self); GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
goto caps_failed; goto caps_failed;
} }
if (!gst_pad_set_caps (GST_BASE_AUDIO_ENCODER_SRC_PAD (self), caps)) { if (!gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (self), caps)) {
gst_caps_unref (caps); gst_caps_unref (caps);
if (buf) if (buf)
gst_omx_port_release_buffer (self->out_port, buf); gst_omx_port_release_buffer (self->out_port, buf);
GST_BASE_AUDIO_ENCODER_STREAM_UNLOCK (self); GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
goto caps_failed; goto caps_failed;
} }
gst_caps_unref (caps); gst_caps_unref (caps);
GST_BASE_AUDIO_ENCODER_STREAM_UNLOCK (self); GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
/* Now get a buffer */ /* Now get a buffer */
if (acq_return != GST_OMX_ACQUIRE_BUFFER_OK) if (acq_return != GST_OMX_ACQUIRE_BUFFER_OK)
@ -468,7 +468,7 @@ gst_omx_audio_enc_loop (GstOMXAudioEnc * self)
GST_DEBUG_OBJECT (self, "Handling buffer: 0x%08x %lu", buf->omx_buf->nFlags, GST_DEBUG_OBJECT (self, "Handling buffer: 0x%08x %lu", buf->omx_buf->nFlags,
buf->omx_buf->nTimeStamp); buf->omx_buf->nTimeStamp);
GST_BASE_AUDIO_ENCODER_STREAM_LOCK (self); GST_AUDIO_ENCODER_STREAM_LOCK (self);
is_eos = ! !(buf->omx_buf->nFlags & OMX_BUFFERFLAG_EOS); is_eos = ! !(buf->omx_buf->nFlags & OMX_BUFFERFLAG_EOS);
if ((buf->omx_buf->nFlags & OMX_BUFFERFLAG_CODECCONFIG) if ((buf->omx_buf->nFlags & OMX_BUFFERFLAG_CODECCONFIG)
@ -476,18 +476,18 @@ gst_omx_audio_enc_loop (GstOMXAudioEnc * self)
GstCaps *caps; GstCaps *caps;
GstBuffer *codec_data; GstBuffer *codec_data;
caps = gst_caps_copy (GST_PAD_CAPS (GST_BASE_AUDIO_ENCODER_SRC_PAD (self))); caps = gst_caps_copy (GST_PAD_CAPS (GST_AUDIO_ENCODER_SRC_PAD (self)));
codec_data = gst_buffer_new_and_alloc (buf->omx_buf->nFilledLen); codec_data = gst_buffer_new_and_alloc (buf->omx_buf->nFilledLen);
memcpy (GST_BUFFER_DATA (codec_data), memcpy (GST_BUFFER_DATA (codec_data),
buf->omx_buf->pBuffer + buf->omx_buf->nOffset, buf->omx_buf->pBuffer + buf->omx_buf->nOffset,
buf->omx_buf->nFilledLen); buf->omx_buf->nFilledLen);
gst_caps_set_simple (caps, "codec_data", GST_TYPE_BUFFER, codec_data, NULL); gst_caps_set_simple (caps, "codec_data", GST_TYPE_BUFFER, codec_data, NULL);
if (!gst_pad_set_caps (GST_BASE_AUDIO_ENCODER_SRC_PAD (self), caps)) { if (!gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (self), caps)) {
gst_caps_unref (caps); gst_caps_unref (caps);
if (buf) if (buf)
gst_omx_port_release_buffer (self->out_port, buf); gst_omx_port_release_buffer (self->out_port, buf);
GST_BASE_AUDIO_ENCODER_STREAM_UNLOCK (self); GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
goto caps_failed; goto caps_failed;
} }
gst_caps_unref (caps); gst_caps_unref (caps);
@ -498,7 +498,7 @@ gst_omx_audio_enc_loop (GstOMXAudioEnc * self)
n_samples = n_samples =
klass->get_num_samples (self, self->out_port, klass->get_num_samples (self, self->out_port,
&GST_BASE_AUDIO_ENCODER (self)->ctx->state, buf); gst_audio_encoder_get_audio_info (GST_AUDIO_ENCODER (self)), buf);
if (buf->omx_buf->nFilledLen > 0) { if (buf->omx_buf->nFilledLen > 0) {
outbuf = gst_buffer_new_and_alloc (buf->omx_buf->nFilledLen); outbuf = gst_buffer_new_and_alloc (buf->omx_buf->nFilledLen);
@ -511,7 +511,7 @@ gst_omx_audio_enc_loop (GstOMXAudioEnc * self)
} }
gst_buffer_set_caps (outbuf, gst_buffer_set_caps (outbuf,
GST_PAD_CAPS (GST_BASE_AUDIO_ENCODER_SRC_PAD (self))); GST_PAD_CAPS (GST_AUDIO_ENCODER_SRC_PAD (self)));
GST_BUFFER_TIMESTAMP (outbuf) = GST_BUFFER_TIMESTAMP (outbuf) =
gst_util_uint64_scale (buf->omx_buf->nTimeStamp, GST_SECOND, gst_util_uint64_scale (buf->omx_buf->nTimeStamp, GST_SECOND,
@ -522,7 +522,7 @@ gst_omx_audio_enc_loop (GstOMXAudioEnc * self)
OMX_TICKS_PER_SECOND); OMX_TICKS_PER_SECOND);
flow_ret = flow_ret =
gst_base_audio_encoder_finish_frame (GST_BASE_AUDIO_ENCODER (self), gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (self),
outbuf, n_samples); outbuf, n_samples);
} }
@ -548,7 +548,7 @@ gst_omx_audio_enc_loop (GstOMXAudioEnc * self)
if (flow_ret != GST_FLOW_OK) if (flow_ret != GST_FLOW_OK)
goto flow_error; goto flow_error;
GST_BASE_AUDIO_ENCODER_STREAM_UNLOCK (self); GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
return; return;
@ -558,9 +558,8 @@ component_error:
("OpenMAX component in error state %s (0x%08x)", ("OpenMAX component in error state %s (0x%08x)",
gst_omx_component_get_last_error_string (self->component), gst_omx_component_get_last_error_string (self->component),
gst_omx_component_get_last_error (self->component))); gst_omx_component_get_last_error (self->component)));
gst_pad_push_event (GST_BASE_AUDIO_ENCODER_SRC_PAD (self), gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (self), gst_event_new_eos ());
gst_event_new_eos ()); gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (self));
gst_pad_pause_task (GST_BASE_AUDIO_ENCODER_SRC_PAD (self));
self->downstream_flow_ret = GST_FLOW_ERROR; self->downstream_flow_ret = GST_FLOW_ERROR;
self->started = FALSE; self->started = FALSE;
return; return;
@ -568,7 +567,7 @@ component_error:
flushing: flushing:
{ {
GST_DEBUG_OBJECT (self, "Flushing -- stopping task"); GST_DEBUG_OBJECT (self, "Flushing -- stopping task");
gst_pad_pause_task (GST_BASE_AUDIO_ENCODER_SRC_PAD (self)); gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (self));
self->downstream_flow_ret = GST_FLOW_WRONG_STATE; self->downstream_flow_ret = GST_FLOW_WRONG_STATE;
self->started = FALSE; self->started = FALSE;
return; return;
@ -578,29 +577,28 @@ flow_error:
if (flow_ret == GST_FLOW_UNEXPECTED) { if (flow_ret == GST_FLOW_UNEXPECTED) {
GST_DEBUG_OBJECT (self, "EOS"); GST_DEBUG_OBJECT (self, "EOS");
gst_pad_push_event (GST_BASE_AUDIO_ENCODER_SRC_PAD (self), gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (self),
gst_event_new_eos ()); gst_event_new_eos ());
gst_pad_pause_task (GST_BASE_AUDIO_ENCODER_SRC_PAD (self)); gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (self));
} else if (flow_ret == GST_FLOW_NOT_LINKED } else if (flow_ret == GST_FLOW_NOT_LINKED
|| flow_ret < GST_FLOW_UNEXPECTED) { || flow_ret < GST_FLOW_UNEXPECTED) {
GST_ELEMENT_ERROR (self, STREAM, FAILED, ("Internal data stream error."), GST_ELEMENT_ERROR (self, STREAM, FAILED, ("Internal data stream error."),
("stream stopped, reason %s", gst_flow_get_name (flow_ret))); ("stream stopped, reason %s", gst_flow_get_name (flow_ret)));
gst_pad_push_event (GST_BASE_AUDIO_ENCODER_SRC_PAD (self), gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (self),
gst_event_new_eos ()); gst_event_new_eos ());
gst_pad_pause_task (GST_BASE_AUDIO_ENCODER_SRC_PAD (self)); gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (self));
} }
self->started = FALSE; self->started = FALSE;
GST_BASE_AUDIO_ENCODER_STREAM_UNLOCK (self); GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
return; return;
} }
reconfigure_error: reconfigure_error:
{ {
GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL), GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL),
("Unable to reconfigure output port")); ("Unable to reconfigure output port"));
gst_pad_push_event (GST_BASE_AUDIO_ENCODER_SRC_PAD (self), gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (self), gst_event_new_eos ());
gst_event_new_eos ()); gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (self));
gst_pad_pause_task (GST_BASE_AUDIO_ENCODER_SRC_PAD (self));
self->downstream_flow_ret = GST_FLOW_NOT_NEGOTIATED; self->downstream_flow_ret = GST_FLOW_NOT_NEGOTIATED;
self->started = FALSE; self->started = FALSE;
return; return;
@ -608,9 +606,8 @@ reconfigure_error:
caps_failed: caps_failed:
{ {
GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL), ("Failed to set caps")); GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL), ("Failed to set caps"));
gst_pad_push_event (GST_BASE_AUDIO_ENCODER_SRC_PAD (self), gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (self), gst_event_new_eos ());
gst_event_new_eos ()); gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (self));
gst_pad_pause_task (GST_BASE_AUDIO_ENCODER_SRC_PAD (self));
self->downstream_flow_ret = GST_FLOW_NOT_NEGOTIATED; self->downstream_flow_ret = GST_FLOW_NOT_NEGOTIATED;
self->started = FALSE; self->started = FALSE;
return; return;
@ -618,7 +615,7 @@ caps_failed:
} }
static gboolean static gboolean
gst_omx_audio_enc_start (GstBaseAudioEncoder * encoder) gst_omx_audio_enc_start (GstAudioEncoder * encoder)
{ {
GstOMXAudioEnc *self; GstOMXAudioEnc *self;
gboolean ret; gboolean ret;
@ -628,14 +625,14 @@ gst_omx_audio_enc_start (GstBaseAudioEncoder * encoder)
self->eos = FALSE; self->eos = FALSE;
self->downstream_flow_ret = GST_FLOW_OK; self->downstream_flow_ret = GST_FLOW_OK;
ret = ret =
gst_pad_start_task (GST_BASE_AUDIO_ENCODER_SRC_PAD (self), gst_pad_start_task (GST_AUDIO_ENCODER_SRC_PAD (self),
(GstTaskFunction) gst_omx_audio_enc_loop, self); (GstTaskFunction) gst_omx_audio_enc_loop, self);
return ret; return ret;
} }
static gboolean static gboolean
gst_omx_audio_enc_stop (GstBaseAudioEncoder * encoder) gst_omx_audio_enc_stop (GstAudioEncoder * encoder)
{ {
GstOMXAudioEnc *self; GstOMXAudioEnc *self;
@ -646,7 +643,7 @@ gst_omx_audio_enc_stop (GstBaseAudioEncoder * encoder)
gst_omx_port_set_flushing (self->in_port, TRUE); gst_omx_port_set_flushing (self->in_port, TRUE);
gst_omx_port_set_flushing (self->out_port, TRUE); gst_omx_port_set_flushing (self->out_port, TRUE);
gst_pad_stop_task (GST_BASE_AUDIO_ENCODER_SRC_PAD (encoder)); gst_pad_stop_task (GST_AUDIO_ENCODER_SRC_PAD (encoder));
if (gst_omx_component_get_state (self->component, 0) > OMX_StateIdle) if (gst_omx_component_get_state (self->component, 0) > OMX_StateIdle)
gst_omx_component_set_state (self->component, OMX_StateIdle); gst_omx_component_set_state (self->component, OMX_StateIdle);
@ -666,8 +663,7 @@ gst_omx_audio_enc_stop (GstBaseAudioEncoder * encoder)
} }
static gboolean static gboolean
gst_omx_audio_enc_set_format (GstBaseAudioEncoder * encoder, gst_omx_audio_enc_set_format (GstAudioEncoder * encoder, GstAudioInfo * info)
GstAudioState * state)
{ {
GstOMXAudioEnc *self; GstOMXAudioEnc *self;
GstOMXAudioEncClass *klass; GstOMXAudioEncClass *klass;
@ -683,10 +679,10 @@ gst_omx_audio_enc_set_format (GstBaseAudioEncoder * encoder,
GST_DEBUG_OBJECT (self, "Setting new caps"); GST_DEBUG_OBJECT (self, "Setting new caps");
/* Set audio encoder base class properties */ /* Set audio encoder base class properties */
encoder->ctx->frame_samples_min = gst_audio_encoder_set_frame_samples_min (encoder,
gst_util_uint64_scale_ceil (OMX_MIN_PCMPAYLOAD_MSEC, gst_util_uint64_scale_ceil (OMX_MIN_PCMPAYLOAD_MSEC,
GST_MSECOND * state->rate, GST_SECOND); GST_MSECOND * info->rate, GST_SECOND));
encoder->ctx->frame_samples_max = 0; gst_audio_encoder_set_frame_samples_max (encoder, 0);
gst_omx_port_get_port_definition (self->in_port, &port_def); gst_omx_port_get_port_definition (self->in_port, &port_def);
@ -714,20 +710,22 @@ gst_omx_audio_enc_set_format (GstBaseAudioEncoder * encoder,
GST_OMX_INIT_STRUCT (&pcm_param); GST_OMX_INIT_STRUCT (&pcm_param);
pcm_param.nPortIndex = self->in_port->index; pcm_param.nPortIndex = self->in_port->index;
pcm_param.nChannels = state->channels; pcm_param.nChannels = info->channels;
pcm_param.eNumData = pcm_param.eNumData =
(state->sign ? OMX_NumericalDataSigned : OMX_NumericalDataUnsigned); ((info->finfo->flags & GST_AUDIO_FORMAT_FLAG_SIGNED) ?
OMX_NumericalDataSigned : OMX_NumericalDataUnsigned);
pcm_param.eEndian = pcm_param.eEndian =
((state->endian == G_LITTLE_ENDIAN) ? OMX_EndianLittle : OMX_EndianBig); ((info->finfo->endianness == G_LITTLE_ENDIAN) ?
OMX_EndianLittle : OMX_EndianBig);
pcm_param.bInterleaved = OMX_TRUE; pcm_param.bInterleaved = OMX_TRUE;
pcm_param.nBitPerSample = state->width; pcm_param.nBitPerSample = info->finfo->width;
pcm_param.nSamplingRate = state->rate; pcm_param.nSamplingRate = info->rate;
pcm_param.ePCMMode = OMX_AUDIO_PCMModeLinear; pcm_param.ePCMMode = OMX_AUDIO_PCMModeLinear;
for (i = 0; i < pcm_param.nChannels; i++) { for (i = 0; i < pcm_param.nChannels; i++) {
OMX_AUDIO_CHANNELTYPE pos; OMX_AUDIO_CHANNELTYPE pos;
switch (state->channel_pos[i]) { switch (info->position[i]) {
case GST_AUDIO_CHANNEL_POSITION_FRONT_MONO: case GST_AUDIO_CHANNEL_POSITION_FRONT_MONO:
case GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER: case GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER:
pos = OMX_AUDIO_ChannelCF; pos = OMX_AUDIO_ChannelCF;
@ -773,7 +771,7 @@ gst_omx_audio_enc_set_format (GstBaseAudioEncoder * encoder,
} }
if (klass->set_format) { if (klass->set_format) {
if (!klass->set_format (self, self->in_port, state)) { if (!klass->set_format (self, self->in_port, info)) {
GST_ERROR_OBJECT (self, "Subclass failed to set the new format"); GST_ERROR_OBJECT (self, "Subclass failed to set the new format");
return FALSE; return FALSE;
} }
@ -821,14 +819,14 @@ gst_omx_audio_enc_set_format (GstBaseAudioEncoder * encoder,
/* Start the srcpad loop again */ /* Start the srcpad loop again */
self->downstream_flow_ret = GST_FLOW_OK; self->downstream_flow_ret = GST_FLOW_OK;
gst_pad_start_task (GST_BASE_AUDIO_ENCODER_SRC_PAD (self), gst_pad_start_task (GST_AUDIO_ENCODER_SRC_PAD (self),
(GstTaskFunction) gst_omx_audio_enc_loop, encoder); (GstTaskFunction) gst_omx_audio_enc_loop, encoder);
return TRUE; return TRUE;
} }
static void static void
gst_omx_audio_enc_flush (GstBaseAudioEncoder * encoder) gst_omx_audio_enc_flush (GstAudioEncoder * encoder)
{ {
GstOMXAudioEnc *self; GstOMXAudioEnc *self;
@ -842,10 +840,10 @@ gst_omx_audio_enc_flush (GstBaseAudioEncoder * encoder)
gst_omx_port_set_flushing (self->out_port, TRUE); gst_omx_port_set_flushing (self->out_port, TRUE);
/* Wait until the srcpad loop is finished */ /* Wait until the srcpad loop is finished */
GST_BASE_AUDIO_ENCODER_STREAM_UNLOCK (self); GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
GST_PAD_STREAM_LOCK (GST_BASE_AUDIO_ENCODER_SRC_PAD (self)); GST_PAD_STREAM_LOCK (GST_AUDIO_ENCODER_SRC_PAD (self));
GST_PAD_STREAM_UNLOCK (GST_BASE_AUDIO_ENCODER_SRC_PAD (self)); GST_PAD_STREAM_UNLOCK (GST_AUDIO_ENCODER_SRC_PAD (self));
GST_BASE_AUDIO_ENCODER_STREAM_LOCK (self); GST_AUDIO_ENCODER_STREAM_LOCK (self);
gst_omx_port_set_flushing (self->in_port, FALSE); gst_omx_port_set_flushing (self->in_port, FALSE);
gst_omx_port_set_flushing (self->out_port, FALSE); gst_omx_port_set_flushing (self->out_port, FALSE);
@ -853,13 +851,12 @@ gst_omx_audio_enc_flush (GstBaseAudioEncoder * encoder)
/* Start the srcpad loop again */ /* Start the srcpad loop again */
self->downstream_flow_ret = GST_FLOW_OK; self->downstream_flow_ret = GST_FLOW_OK;
self->eos = FALSE; self->eos = FALSE;
gst_pad_start_task (GST_BASE_AUDIO_ENCODER_SRC_PAD (self), gst_pad_start_task (GST_AUDIO_ENCODER_SRC_PAD (self),
(GstTaskFunction) gst_omx_audio_enc_loop, encoder); (GstTaskFunction) gst_omx_audio_enc_loop, encoder);
} }
static GstFlowReturn static GstFlowReturn
gst_omx_audio_enc_handle_frame (GstBaseAudioEncoder * encoder, gst_omx_audio_enc_handle_frame (GstAudioEncoder * encoder, GstBuffer * inbuf)
GstBuffer * inbuf)
{ {
GstOMXAcquireBufferReturn acq_ret = GST_OMX_ACQUIRE_BUFFER_ERROR; GstOMXAcquireBufferReturn acq_ret = GST_OMX_ACQUIRE_BUFFER_ERROR;
GstOMXAudioEnc *self; GstOMXAudioEnc *self;
@ -893,9 +890,9 @@ gst_omx_audio_enc_handle_frame (GstBaseAudioEncoder * encoder,
/* Make sure to release the base class stream lock, otherwise /* Make sure to release the base class stream lock, otherwise
* _loop() can't call _finish_frame() and we might block forever * _loop() can't call _finish_frame() and we might block forever
* because no input buffers are released */ * because no input buffers are released */
GST_BASE_AUDIO_ENCODER_STREAM_UNLOCK (self); GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
acq_ret = gst_omx_port_acquire_buffer (self->in_port, &buf); acq_ret = gst_omx_port_acquire_buffer (self->in_port, &buf);
GST_BASE_AUDIO_ENCODER_STREAM_LOCK (self); GST_AUDIO_ENCODER_STREAM_LOCK (self);
if (acq_ret == GST_OMX_ACQUIRE_BUFFER_ERROR) { if (acq_ret == GST_OMX_ACQUIRE_BUFFER_ERROR) {
goto component_error; goto component_error;
@ -990,7 +987,7 @@ reconfigure_error:
} }
static gboolean static gboolean
gst_omx_audio_enc_event (GstBaseAudioEncoder * encoder, GstEvent * event) gst_omx_audio_enc_event (GstAudioEncoder * encoder, GstEvent * event)
{ {
GstOMXAudioEnc *self; GstOMXAudioEnc *self;
@ -1012,7 +1009,7 @@ gst_omx_audio_enc_event (GstBaseAudioEncoder * encoder, GstEvent * event)
/* Make sure to release the base class stream lock, otherwise /* Make sure to release the base class stream lock, otherwise
* _loop() can't call _finish_frame() and we might block forever * _loop() can't call _finish_frame() and we might block forever
* because no input buffers are released */ * because no input buffers are released */
GST_BASE_AUDIO_ENCODER_STREAM_UNLOCK (self); GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
/* Send an EOS buffer to the component and let the base /* Send an EOS buffer to the component and let the base
* class drop the EOS event. We will send it later when * class drop the EOS event. We will send it later when
@ -1026,7 +1023,7 @@ gst_omx_audio_enc_event (GstBaseAudioEncoder * encoder, GstEvent * event)
GST_ERROR_OBJECT (self, "Failed to acquire buffer for EOS: %d", acq_ret); GST_ERROR_OBJECT (self, "Failed to acquire buffer for EOS: %d", acq_ret);
} }
GST_BASE_AUDIO_ENCODER_STREAM_LOCK (self); GST_AUDIO_ENCODER_STREAM_LOCK (self);
return FALSE; return FALSE;
} }
@ -1057,14 +1054,14 @@ gst_omx_audio_enc_drain (GstOMXAudioEnc * self)
/* Make sure to release the base class stream lock, otherwise /* Make sure to release the base class stream lock, otherwise
* _loop() can't call _finish_frame() and we might block forever * _loop() can't call _finish_frame() and we might block forever
* because no input buffers are released */ * because no input buffers are released */
GST_BASE_AUDIO_ENCODER_STREAM_UNLOCK (self); GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
/* Send an EOS buffer to the component and let the base /* Send an EOS buffer to the component and let the base
* class drop the EOS event. We will send it later when * class drop the EOS event. We will send it later when
* the EOS buffer arrives on the output port. */ * the EOS buffer arrives on the output port. */
acq_ret = gst_omx_port_acquire_buffer (self->in_port, &buf); acq_ret = gst_omx_port_acquire_buffer (self->in_port, &buf);
if (acq_ret != GST_OMX_ACQUIRE_BUFFER_OK) { if (acq_ret != GST_OMX_ACQUIRE_BUFFER_OK) {
GST_BASE_AUDIO_ENCODER_STREAM_LOCK (self); GST_AUDIO_ENCODER_STREAM_LOCK (self);
GST_ERROR_OBJECT (self, "Failed to acquire buffer for draining: %d", GST_ERROR_OBJECT (self, "Failed to acquire buffer for draining: %d",
acq_ret); acq_ret);
return GST_FLOW_ERROR; return GST_FLOW_ERROR;
@ -1078,7 +1075,7 @@ gst_omx_audio_enc_drain (GstOMXAudioEnc * self)
g_cond_wait (self->drain_cond, self->drain_lock); g_cond_wait (self->drain_cond, self->drain_lock);
GST_DEBUG_OBJECT (self, "Drained component"); GST_DEBUG_OBJECT (self, "Drained component");
g_mutex_unlock (self->drain_lock); g_mutex_unlock (self->drain_lock);
GST_BASE_AUDIO_ENCODER_STREAM_LOCK (self); GST_AUDIO_ENCODER_STREAM_LOCK (self);
self->started = FALSE; self->started = FALSE;

12
omx/gstomxaudioenc.h
View file

@ -22,7 +22,7 @@
#define __GST_OMX_AUDIO_ENC_H__ #define __GST_OMX_AUDIO_ENC_H__
#include <gst/gst.h> #include <gst/gst.h>
#include "gstbaseaudioencoder.h" #include <gst/audio/gstaudioencoder.h>
#include "gstomx.h" #include "gstomx.h"
@ -46,7 +46,7 @@ typedef struct _GstOMXAudioEncClass GstOMXAudioEncClass;
struct _GstOMXAudioEnc struct _GstOMXAudioEnc
{ {
GstBaseAudioEncoder parent; GstAudioEncoder parent;
/* < protected > */ /* < protected > */
GstOMXCore *core; GstOMXCore *core;
@ -72,7 +72,7 @@ struct _GstOMXAudioEnc
struct _GstOMXAudioEncClass struct _GstOMXAudioEncClass
{ {
GstBaseAudioEncoderClass parent_class; GstAudioEncoderClass parent_class;
const gchar *core_name; const gchar *core_name;
const gchar *component_name; const gchar *component_name;
@ -85,9 +85,9 @@ struct _GstOMXAudioEncClass
guint64 hacks; guint64 hacks;
gboolean (*set_format) (GstOMXAudioEnc * self, GstOMXPort * port, GstAudioState * state); gboolean (*set_format) (GstOMXAudioEnc * self, GstOMXPort * port, GstAudioInfo * info);
GstCaps *(*get_caps) (GstOMXAudioEnc * self, GstOMXPort * port, GstAudioState * state); GstCaps *(*get_caps) (GstOMXAudioEnc * self, GstOMXPort * port, GstAudioInfo * info);
guint (*get_num_samples) (GstOMXAudioEnc * self, GstOMXPort * port, GstAudioState * state, GstOMXBuffer * buffer); guint (*get_num_samples) (GstOMXAudioEnc * self, GstOMXPort * port, GstAudioInfo * info, GstOMXBuffer * buffer);
}; };
GType gst_omx_audio_enc_get_type (void); GType gst_omx_audio_enc_get_type (void);