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baseaudiodecoder: initial version
This commit is contained in:
Mark Nauwelaerts
Tim-Philipp Müller
parent
87409f2587
commit
1dbbe7c89d
2 changed files with 1589 additions and 849 deletions
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@ -1,6 +1,9 @@
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/* GStreamer
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* Copyright (C) 2009 Igalia S.L.
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* Author: Iago Toral Quiroga <itoral@igalia.com>
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* Copyright (C) 2011 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>.
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* Copyright (C) 2011 Nokia Corporation. All rights reserved.
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* Contact: Stefan Kost <stefan.kost@nokia.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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@ -27,6 +30,7 @@
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#endif
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#include <gst/gst.h>
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#include <gst/audio/gstbaseaudioutils.h>
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#include <gst/base/gstadapter.h>
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G_BEGIN_DECLS
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@ -73,59 +77,46 @@ G_BEGIN_DECLS
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*/
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#define GST_BASE_AUDIO_DECODER_SINK_PAD(obj) (((GstBaseAudioDecoder *) (obj))->sinkpad)
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/**
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* GST_BASE_AUDIO_DECODER_INPUT_ADAPTER:
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* @obj: base audio codec instance
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*
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* Gives the pointer to the input #GstAdapter object of the element.
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*/
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#define GST_BASE_AUDIO_DECODER_INPUT_ADAPTER(obj) (((GstBaseAudioDecoder *) (obj))->input_adapter)
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/**
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* GST_BASE_AUDIO_DECODER_OUTPUT_ADAPTER:
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* @obj: base audio codec instance
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*
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* Gives the pointer to the output #GstAdapter object of the element.
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*/
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#define GST_BASE_AUDIO_DECODER_OUTPUT_ADAPTER(obj) (((GstBaseAudioDecoder *) (obj))->output_adapter)
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typedef struct _GstBaseAudioDecoder GstBaseAudioDecoder;
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typedef struct _GstBaseAudioDecoderClass GstBaseAudioDecoderClass;
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typedef struct _GstAudioState GstAudioState;
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struct _GstAudioState
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{
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gint channels;
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gint rate;
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gint bytes_per_sample;
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gint sample_depth;
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gint frame_size;
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GstSegment segment;
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typedef struct _GstBaseAudioDecoderPrivate GstBaseAudioDecoderPrivate;
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typedef struct _GstBaseAudioDecoderContext GstBaseAudioDecoderContext;
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/**
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* GstBaseAudioDecoderContext:
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* @state: a #GstAudioState describing input audio format
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* @eos: no (immediate) subsequent data in stream
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* @sync: stream parsing in sync
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* @delay: number of frames pending decoding (typically at least 1 for current)
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* @do_plc: whether subclass is prepared to handle (packet) loss concealment
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* @min_latency: min latency of element
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* @max_latency: max latency of element
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* @lookahead: decoder lookahead (in units of input rate samples)
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*
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* Transparent #GstBaseAudioEncoderContext data structure.
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*/
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struct _GstBaseAudioDecoderContext {
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/* input */
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/* (output) audio format */
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GstAudioState state;
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/* parsing state */
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gboolean eos;
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gboolean sync;
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/* misc */
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gint delay;
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/* output */
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gboolean do_plc;
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/* MT-protected (with LOCK) */
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GstClockTime min_latency;
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GstClockTime max_latency;
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};
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/**
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* GstBaseAudioDecoder:
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* @element: the parent element.
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* @caps_set: whether caps have been set on the codec's source pad.
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* @sinkpad: the sink pad.
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* @srcpad: the source pad.
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* @input_adapter: the input adapter that will be filled with the input buffers.
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* @output_adapter: the output adapter. Subclasses will read from the input
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* adapter, process the data and fill the output adapter with the result.
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* @input_buffer_size: The minimum amount of data that should be present on the
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* input adapter for the codec to process it.
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* @output_buffer_size: The minimum amount of data that should be present on the
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* output adapter for the codec to push buffers out.
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* @bytes_in: total bytes that have been received.
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* @bytes_out: total bytes that have been pushed out.
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* @discont: whether the next buffer to push represents a discontinuity in the
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* stream.
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* @state: Audio stream information. See #GstAudioState for details.
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* @codec_data: The codec data.
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* @started: Whether the codec has been started and is ready to process data
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* or not.
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* @first_ts: timestamp of the first buffer in the input adapter.
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* @last_ts: timestamp of the last buffer in the input adapter.
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*
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* The opaque #GstBaseAudioDecoder data structure.
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*/
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@ -133,86 +124,97 @@ struct _GstBaseAudioDecoder
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{
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GstElement element;
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/*< private >*/
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gboolean caps_set;
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/*< protected >*/
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GstPad *sinkpad;
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GstPad *srcpad;
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GstAdapter *input_adapter;
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GstAdapter *output_adapter;
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guint input_buffer_size;
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guint output_buffer_size;
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guint64 bytes_in;
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guint64 bytes_out;
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gboolean discont;
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GstAudioState state;
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GstBuffer *codec_data;
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gboolean started;
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/* source and sink pads */
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GstPad *sinkpad;
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GstPad *srcpad;
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guint64 first_ts;
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guint64 last_ts;
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/* MT-protected (with STREAM_LOCK) */
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GstSegment segment;
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GstBaseAudioDecoderContext *ctx;
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/* properties */
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GstClockTime latency;
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GstClockTime tolerance;
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gboolean plc;
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/*< private >*/
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GstBaseAudioDecoderPrivate *priv;
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gpointer _gst_reserved[GST_PADDING_LARGE];
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};
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/**
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* GstBaseAudioDecoderClass:
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* @parent_class: Element parent class
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* @start: Start processing. Ideal for opening resources in the subclass
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* @stop: Stop processing. Subclasses should use this to close resources.
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* @reset: Resets the codec. Called on discontinuities, etc.
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* @event: Override this to handle events arriving on the sink pad.
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* @handle_discont: Override to be notified on discontinuities.
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* @flush_input: Subclasses may implement this to flush the input adapter,
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* processing any data present in it and filling the output adapter with the
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* result. This could be necessary if it is possible for the codec to
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* receive an end-of-stream event before all the data in the input
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* adapter has been processed.
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* @flush_output: Subclasses may implement this to flush the output adapter,
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* pushing buffers out through the codec's source pad when the end-of-stream
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* event is received and there is data waiting to be processed in the
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* adapters.
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* @process_data: Subclasses must implement this. They should read from the
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* input adapter, encode/decode the data present in it and fill the
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* output adapter with the result.
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* @push_data: Normally, #GstBaseAudioDecoder will handle pushing buffers out.
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* However, it is possible for developers to take control of when and how
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* buffers are pushed out by overriding this method. If subclasses provide
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* an implementation, #GstBaseAudioDecoder will not push any buffers,
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* instead, whenever there is data on the output adapter, it will call this
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* method on the subclass, which would be the sole responsible for
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* pushing the buffers out when appropriate.
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* @negotiate_src_caps: Subclasses can implement this method to provide
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* appropriate caps to be set on the codec's source pad. If they don't
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* provide this, they will be responsible for calling
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* gst_base_audio_decoder_set_src_caps when appropriate.
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* @start: Optional.
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* Called when the element starts processing.
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* Allows opening external resources.
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* @stop: Optional.
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* Called when the element stops processing.
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* Allows closing external resources.
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* @set_format: Notifies subclass of incoming data format (caps).
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* @parse: Optional.
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* Allows chopping incoming data into manageable units (frames)
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* for subsequent decoding. This division is at subclass
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* discretion and may or may not correspond to 1 (or more)
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* frames as defined by audio format.
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* @handle_frame: Provides input data (or NULL to clear any remaining data)
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* to subclass. Input data ref management is performed by
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* base class, subclass should not care or intervene.
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* @flush: Optional.
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* Instructs subclass to clear any codec caches and discard
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* any pending samples and not yet returned encoded data.
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* @hard indicates whether a FLUSH is being processed,
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* or otherwise a DISCONT (or conceptually similar).
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* @event: Optional.
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* Event handler on the sink pad. This function should return
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* TRUE if the event was handled and should be discarded
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* (i.e. not unref'ed).
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* @pre_push: Optional.
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* Called just prior to pushing (encoded data) buffer downstream.
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* Subclass has full discretionary access to buffer,
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* and a not OK flow return will abort downstream pushing.
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*
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* Subclasses can override any of the available virtual methods or not, as
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* needed. At minimum @handle_frame (and likely @set_format) needs to be
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* overridden.
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*/
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struct _GstBaseAudioDecoderClass
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{
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GstElementClass parent_class;
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gboolean (*start) (GstBaseAudioDecoder *codec);
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gboolean (*stop) (GstBaseAudioDecoder *codec);
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gboolean (*reset) (GstBaseAudioDecoder *codec);
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/*< public >*/
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/* virtual methods for subclasses */
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GstFlowReturn (*event) (GstBaseAudioDecoder *codec, GstEvent *event);
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void (*handle_discont) (GstBaseAudioDecoder *codec, GstBuffer *buffer);
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gboolean (*flush_input) (GstBaseAudioDecoder *codec);
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gboolean (*flush_output) (GstBaseAudioDecoder *codec);
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GstFlowReturn (*process_data) (GstBaseAudioDecoder *codec);
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GstFlowReturn (*push_data) (GstBaseAudioDecoder *codec);
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GstCaps * (*negotiate_src_caps) (GstBaseAudioDecoder *codec,
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GstCaps *sink_caps);
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gboolean (*start) (GstBaseAudioDecoder *dec);
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gboolean (*stop) (GstBaseAudioDecoder *dec);
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gboolean (*set_format) (GstBaseAudioDecoder *dec,
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GstCaps *caps);
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GstFlowReturn (*parse) (GstBaseAudioDecoder *dec,
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GstAdapter *adapter,
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gint *offset, gint *length);
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GstFlowReturn (*handle_frame) (GstBaseAudioDecoder *dec,
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GstBuffer *buffer);
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void (*flush) (GstBaseAudioDecoder *dec, gboolean hard);
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GstFlowReturn (*pre_push) (GstBaseAudioDecoder *dec,
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GstBuffer **buffer);
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gboolean (*event) (GstBaseAudioDecoder *dec,
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GstEvent *event);
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/*< private >*/
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gpointer _gst_reserved[GST_PADDING_LARGE];
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};
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GstFlowReturn gst_base_audio_decoder_finish_frame (GstBaseAudioDecoder * dec,
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GstBuffer * buf, gint frames);
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GType gst_base_audio_decoder_get_type (void);
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gboolean gst_base_audio_decoder_reset (GstBaseAudioDecoder *codec);
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gboolean gst_base_audio_decoder_stop (GstBaseAudioDecoder *codec);
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gboolean gst_base_audio_decoder_start (GstBaseAudioDecoder *codec);
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gboolean gst_base_audio_decoder_flush (GstBaseAudioDecoder *codec);
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gboolean gst_base_audio_decoder_set_src_caps (GstBaseAudioDecoder *codec,
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GstCaps *caps);
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GstFlowReturn gst_base_audio_decoder_push_buffer (GstBaseAudioDecoder *codec,
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GstBuffer *buffer);
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G_END_DECLS
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