From 80241fde8d3d50e30a0d9287a772789cc0e2ab49 Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Thu, 27 Jan 2011 16:52:50 +0100 Subject: [PATCH] audioencoders: baseaudioencoder and ported encoders --- gst-libs/gst/audio/gstbaseaudioencoder.c | 1593 ++++++++++++++++++++++ gst-libs/gst/audio/gstbaseaudioencoder.h | 251 ++++ 2 files changed, 1844 insertions(+) create mode 100644 gst-libs/gst/audio/gstbaseaudioencoder.c create mode 100644 gst-libs/gst/audio/gstbaseaudioencoder.h diff --git a/gst-libs/gst/audio/gstbaseaudioencoder.c b/gst-libs/gst/audio/gstbaseaudioencoder.c new file mode 100644 index 0000000000..f539ebd176 --- /dev/null +++ b/gst-libs/gst/audio/gstbaseaudioencoder.c @@ -0,0 +1,1593 @@ +/* GStreamer + * Copyright (C) 2011 Mark Nauwelaerts . + * Copyright (C) 2011 Nokia Corporation. All rights reserved. + * Contact: Stefan Kost + * + * 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. + */ + +/** + * SECTION:gstbaseaudioencoder + * @short_description: Base class for audio encoders + * @see_also: #GstBaseTransform + * + * This base class is for audio encoders turning raw audio samples into + * encoded audio data. + * + * GstBaseAudioEncoder and subclass should cooperate as follows. + * + * + * Configuration + * + * Initially, GstBaseAudioEncoder calls @start when the encoder element + * is activated, which allows subclass to perform any global setup. + * + * + * GstBaseAudioEncoder calls @set_format to inform subclass of the format + * of input audio data that it is about to receive. Subclass should + * setup for encoding and configure various base class context parameters + * appropriately, notably those directing desired input data handling. + * While unlikely, it might be called more than once, if changing input + * parameters require reconfiguration. + * + * + * GstBaseAudioEncoder calls @stop at end of all processing. + * + * + * + * As of configuration stage, and throughout processing, GstBaseAudioEncoder + * provides a GstBaseAudioEncoderContext that provides required context, + * e.g. describing the format of input audio data. + * Conversely, subclass can and should configure context to inform + * base class of its expectation w.r.t. buffer handling. + * + * + * Data processing + * + * Base class gathers input sample data (as directed by the context's + * frame_samples and frame_max) and provides this to subclass' @handle_frame. + * + * + * If codec processing results in encoded data, subclass should call + * @gst_base_audio_encoder_finish_frame to have encoded data pushed + * downstream. Alternatively, it might also call to indicate dropped + * (non-encoded) samples. + * + * + * Just prior to actually pushing a buffer downstream, + * it is passed to @pre_push. + * + * + * During the parsing process GstBaseAudioEncoderClass will handle both + * srcpad and sinkpad events. Sink events will be passed to subclass + * if @event callback has been provided. + * + * + * + * + * Shutdown phase + * + * GstBaseAudioEncoder class calls @stop to inform the subclass that data + * parsing will be stopped. + * + * + * + * + * + * Subclass is responsible for providing pad template caps for + * source and sink pads. The pads need to be named "sink" and "src". It also + * needs to set the fixed caps on srcpad, when the format is ensured. This + * is typically when base class calls subclass' @set_format function, though + * it might be delayed until calling @gst_base_audio_encoder_finish_frame. + * + * In summary, above process should have subclass concentrating on + * codec data processing while leaving other matters to base class, + * such as most notably timestamp handling. While it may exert more control + * in this area (see e.g. @pre_push), it is very much not recommended. + * + * In particular, base class will either favor tracking upstream timestamps + * (at the possible expense of jitter) or aim to arrange for a perfect stream of + * output timestamps, depending on + * perfect-ts. + * However, in the latter case, the input may not be so perfect or ideal, which + * is handled as follows. An input timestamp is compared with the expected + * timestamp as dictated by input sample stream and if the deviation is less + * than tolerance, + * the deviation is discarded. Otherwise, it is considered + * a discontuinity and subsequent output timestamp is resynced to the + * new position after performing configured discontinuity processing. + * In the non-perfect-ts case, an upstream variation exceeding tolerance + * only leads to marking DISCONT on subsequent outgoing (while timestamps + * are adjusted to upstream regardless of variation). + * While DISCONT is also marked in the perfect-ts case, this one optionally + * (see hard-resync) + * performs some additional steps, such as clipping of (early) input samples + * or draining all currently remaining input data, depending on the direction + * of the discontuinity. + * + * Things that subclass need to take care of: + * + * Provide pad templates + * + * Set source pad caps when appropriate + * + * + * Inform base class of buffer processing needs using context's + * frame_samples and frame_bytes. + * + * + * Set user-configurable properties to sane defaults for format and + * implementing codec at hand, e.g. those controlling timestamp behaviour + * and discontinuity processing. + * + * + * Accept data in @handle_frame and provide encoded results to + * @gst_base_audio_encoder_finish_frame. + * + * + * + */ + +#ifdef HAVE_CONFIG_H +# include "config.h" +#endif + +#include +#include + +#include "gstbaseaudioencoder.h" + +#include + +GST_DEBUG_CATEGORY_STATIC (gst_base_audio_encoder_debug); +#define GST_CAT_DEFAULT gst_base_audio_encoder_debug + +#define GST_BASE_AUDIO_ENCODER_GET_PRIVATE(obj) \ + (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_BASE_AUDIO_ENCODER, \ + GstBaseAudioEncoderPrivate)) + +enum +{ + PROP_0, + PROP_PERFECT_TS, + PROP_GRANULE, + PROP_HARD_RESYNC, + PROP_TOLERANCE +}; + +#define DEFAULT_PERFECT_TS FALSE +#define DEFAULT_GRANULE FALSE +#define DEFAULT_HARD_RESYNC FALSE +#define DEFAULT_TOLERANCE 40000000 + +struct _GstBaseAudioEncoderPrivate +{ + /* activation status */ + gboolean active; + + /* input base/first ts as basis for output ts; + * kept nearly constant for perfect_ts, + * otherwise resyncs to upstream ts */ + GstClockTime base_ts; + /* corresponding base granulepos */ + gint64 base_gp; + /* input samples processed and sent downstream so far (w.r.t. base_ts) */ + guint64 samples; + + /* currently collected sample data */ + GstAdapter *adapter; + /* offset in adapter up to which already supplied to encoder */ + gint offset; + /* collected encoded data */ + GstAdapter *adapter_out; + /* (estimated) samples (w.r.t. input rate) represented in adapter_out */ + gint samples_out; + /* mark outgoing discont */ + gboolean discont; + /* to guess duration of drained data */ + GstClockTime last_duration; + + /* subclass provided data in processing round */ + gboolean got_data; + /* subclass gave all it could already */ + gboolean drained; + /* subclass currently being forcibly drained */ + gboolean force; + + /* MT safe latency; taken from ctx */ + GstClockTime min_latency; + GstClockTime max_latency; + /* output bps estimatation */ + /* global in samples seen */ + guint64 samples_in; + /* global bytes sent out */ + guint64 bytes_out; + + /* context storage */ + GstBaseAudioEncoderContext ctx; +}; + + +static GstElementClass *parent_class = NULL; + +static void gst_base_audio_encoder_class_init (GstBaseAudioEncoderClass * + klass); +static void gst_base_audio_encoder_init (GstBaseAudioEncoder * parse, + GstBaseAudioEncoderClass * klass); + +GType +gst_base_audio_encoder_get_type (void) +{ + static GType base_audio_encoder_type = 0; + + if (!base_audio_encoder_type) { + static const GTypeInfo base_audio_encoder_info = { + sizeof (GstBaseAudioEncoderClass), + (GBaseInitFunc) NULL, + (GBaseFinalizeFunc) NULL, + (GClassInitFunc) gst_base_audio_encoder_class_init, + NULL, + NULL, + sizeof (GstBaseAudioEncoder), + 0, + (GInstanceInitFunc) gst_base_audio_encoder_init, + }; + const GInterfaceInfo preset_interface_info = { + NULL, /* interface_init */ + NULL, /* interface_finalize */ + NULL /* interface_data */ + }; + + base_audio_encoder_type = g_type_register_static (GST_TYPE_ELEMENT, + "GstBaseAudioEncoder", &base_audio_encoder_info, G_TYPE_FLAG_ABSTRACT); + + g_type_add_interface_static (base_audio_encoder_type, GST_TYPE_PRESET, + &preset_interface_info); + } + return base_audio_encoder_type; +} + +static void gst_base_audio_encoder_finalize (GObject * object); +static void gst_base_audio_encoder_reset (GstBaseAudioEncoder * enc, + gboolean full); + +static void gst_base_audio_encoder_set_property (GObject * object, + guint prop_id, const GValue * value, GParamSpec * pspec); +static void gst_base_audio_encoder_get_property (GObject * object, + guint prop_id, GValue * value, GParamSpec * pspec); + +static gboolean gst_base_audio_encoder_sink_activate_push (GstPad * pad, + gboolean active); + +static gboolean gst_base_audio_encoder_sink_event (GstPad * pad, + GstEvent * event); +static gboolean gst_base_audio_encoder_sink_setcaps (GstPad * pad, + GstCaps * caps); +static GstFlowReturn gst_base_audio_encoder_chain (GstPad * pad, + GstBuffer * buffer); +static gboolean gst_base_audio_encoder_src_query (GstPad * pad, + GstQuery * query); +static gboolean gst_base_audio_encoder_sink_query (GstPad * pad, + GstQuery * query); +static const GstQueryType *gst_base_audio_encoder_get_query_types (GstPad * + pad); +static GstCaps *gst_base_audio_encoder_sink_getcaps (GstPad * pad); + + +static void +gst_base_audio_encoder_class_init (GstBaseAudioEncoderClass * klass) +{ + GObjectClass *gobject_class; + + gobject_class = G_OBJECT_CLASS (klass); + parent_class = g_type_class_peek_parent (klass); + + GST_DEBUG_CATEGORY_INIT (gst_base_audio_encoder_debug, "baseaudioencoder", 0, + "baseaudioencoder element"); + + g_type_class_add_private (klass, sizeof (GstBaseAudioEncoderPrivate)); + + gobject_class->set_property = gst_base_audio_encoder_set_property; + gobject_class->get_property = gst_base_audio_encoder_get_property; + + gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_base_audio_encoder_finalize); + + /* properties */ + g_object_class_install_property (gobject_class, PROP_PERFECT_TS, + g_param_spec_boolean ("perfect-ts", "perfect-ts", + "Favour perfect timestamps over tracking upstream timestamps", + DEFAULT_PERFECT_TS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + g_object_class_install_property (gobject_class, PROP_GRANULE, + g_param_spec_boolean ("granule", "granule", + "Apply granule semantics to buffer metadata (implies perfect-ts)", + DEFAULT_GRANULE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); + g_object_class_install_property (gobject_class, PROP_HARD_RESYNC, + g_param_spec_boolean ("hard-resync", "hard-resync", + "Perform clipping and sample flushing upon discontinuity", + DEFAULT_HARD_RESYNC, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + g_object_class_install_property (gobject_class, PROP_TOLERANCE, + g_param_spec_int64 ("tolerance", "tolerance", + "Consider discontinuity if timestamp jitter/imperfection exceeds tolerance", + 0, G_MAXINT64, DEFAULT_TOLERANCE, + G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); +} + +static void +gst_base_audio_encoder_init (GstBaseAudioEncoder * enc, + GstBaseAudioEncoderClass * bclass) +{ + GstPadTemplate *pad_template; + + GST_DEBUG_OBJECT (enc, "gst_base_audio_encoder_init"); + + enc->priv = GST_BASE_AUDIO_ENCODER_GET_PRIVATE (enc); + + /* only push mode supported */ + pad_template = + gst_element_class_get_pad_template (GST_ELEMENT_CLASS (bclass), "sink"); + g_return_if_fail (pad_template != NULL); + enc->sinkpad = gst_pad_new_from_template (pad_template, "sink"); + gst_pad_set_event_function (enc->sinkpad, + GST_DEBUG_FUNCPTR (gst_base_audio_encoder_sink_event)); + gst_pad_set_setcaps_function (enc->sinkpad, + GST_DEBUG_FUNCPTR (gst_base_audio_encoder_sink_setcaps)); + gst_pad_set_getcaps_function (enc->sinkpad, + GST_DEBUG_FUNCPTR (gst_base_audio_encoder_sink_getcaps)); + gst_pad_set_query_function (enc->sinkpad, + GST_DEBUG_FUNCPTR (gst_base_audio_encoder_sink_query)); + gst_pad_set_chain_function (enc->sinkpad, + GST_DEBUG_FUNCPTR (gst_base_audio_encoder_chain)); + gst_pad_set_activatepush_function (enc->sinkpad, + GST_DEBUG_FUNCPTR (gst_base_audio_encoder_sink_activate_push)); + gst_element_add_pad (GST_ELEMENT (enc), enc->sinkpad); + + GST_DEBUG_OBJECT (enc, "sinkpad created"); + + /* and we don't mind upstream traveling stuff that much ... */ + pad_template = + gst_element_class_get_pad_template (GST_ELEMENT_CLASS (bclass), "src"); + g_return_if_fail (pad_template != NULL); + enc->srcpad = gst_pad_new_from_template (pad_template, "src"); + gst_pad_set_query_function (enc->srcpad, + GST_DEBUG_FUNCPTR (gst_base_audio_encoder_src_query)); + gst_pad_set_query_type_function (enc->srcpad, + GST_DEBUG_FUNCPTR (gst_base_audio_encoder_get_query_types)); + gst_pad_use_fixed_caps (enc->srcpad); + gst_element_add_pad (GST_ELEMENT (enc), enc->srcpad); + GST_DEBUG_OBJECT (enc, "src created"); + + enc->priv->adapter = gst_adapter_new (); + enc->priv->adapter_out = gst_adapter_new (); + enc->ctx = &enc->priv->ctx; + + /* property default */ + enc->perfect_ts = DEFAULT_PERFECT_TS; + enc->hard_resync = DEFAULT_HARD_RESYNC; + enc->tolerance = DEFAULT_TOLERANCE; + + /* init state */ + gst_base_audio_encoder_reset (enc, TRUE); + GST_DEBUG_OBJECT (enc, "init ok"); +} + +static void +gst_base_audio_encoder_reset (GstBaseAudioEncoder * enc, gboolean full) +{ + GST_OBJECT_LOCK (enc); + + if (full) { + enc->priv->active = FALSE; + enc->priv->samples_in = 0; + enc->priv->bytes_out = 0; + memset (enc->ctx, 0, sizeof (enc->ctx)); + enc->ctx->state.bpf = 0; + enc->ctx->state.rate = 0; + enc->ctx->min_latency = 0; + enc->ctx->max_latency = 0; + g_free (enc->ctx->state.channel_pos); + enc->ctx->state.channel_pos = NULL; + } + + gst_segment_init (&enc->segment, GST_FORMAT_TIME); + + gst_adapter_clear (enc->priv->adapter); + gst_adapter_clear (enc->priv->adapter_out); + enc->priv->got_data = FALSE; + enc->priv->drained = TRUE; + enc->priv->offset = 0; + enc->priv->base_ts = GST_CLOCK_TIME_NONE; + enc->priv->base_gp = -1; + enc->priv->samples = 0; + enc->priv->samples_out = 0; + enc->priv->discont = FALSE; + + GST_OBJECT_UNLOCK (enc); +} + +static void +gst_base_audio_encoder_finalize (GObject * object) +{ + GstBaseAudioEncoder *enc = GST_BASE_AUDIO_ENCODER (object); + + g_object_unref (enc->priv->adapter); + g_object_unref (enc->priv->adapter_out); + + G_OBJECT_CLASS (parent_class)->finalize (object); +} + +/** gst_base_audio_encoder_finish_frame: + * @enc: a #GstBaseAudioEncoder + * @buffer: encoded data + * @samples: number of samples (per channel) represented by encoded data + * + * Collects encoded data and/or pushes encoded data downstream. + * Source pad caps must be set when this is called. Depending on the nature + * of the (framing of) the format, subclass can decide whether to push + * encoded data directly or to collect various "frames" in a single buffer. + * Note that the latter behaviour is recommended whenever the format is allowed, + * as it incurs no additional latency and avoids otherwise generating a + * a multitude of (small) output buffers. If not explicitly pushed, + * any available encoded data is pushed at the end of each processing cycle, + * i.e. which encodes as much data as available input data allows. + * + * If @samples < 0, then best estimate is all samples provided to encoder + * (subclass) so far. @buf may be NULL, in which case next number of @samples + * are considered discarded, e.g. as a result of discontinuous transmission, + * and a discontinuity is marked (note that @buf == NULL => push == TRUE). + * + * Returns: a #GstFlowReturn that should be escalated to caller (of caller) + */ +GstFlowReturn +gst_base_audio_encoder_finish_frame (GstBaseAudioEncoder * enc, GstBuffer * buf, + gint samples) +{ + GstBaseAudioEncoderClass *klass; + GstBaseAudioEncoderPrivate *priv; + GstBaseAudioEncoderContext *ctx; + GstFlowReturn ret = GST_FLOW_OK; + gint av; + + klass = GST_BASE_AUDIO_ENCODER_GET_CLASS (enc); + priv = enc->priv; + ctx = enc->ctx; + + /* subclass should know what it is producing by now */ + g_return_val_if_fail (GST_PAD_CAPS (enc->srcpad) != NULL, GST_FLOW_ERROR); + /* subclass should not hand us no data */ + g_return_val_if_fail (buf == NULL || GST_BUFFER_SIZE (buf) > 0, + GST_FLOW_ERROR); + + GST_LOG_OBJECT (enc, "accepting %d bytes encoded data as %d samples", + buf ? GST_BUFFER_SIZE (buf) : -1, samples); + + /* mark subclass still alive and providing */ + priv->got_data = TRUE; + + /* remove corresponding samples from input */ + if (samples < 0) + samples = (enc->priv->offset / ctx->state.bpf); + + if (G_LIKELY (samples)) { + /* track upstream ts at output collection start if so configured */ + if (!enc->perfect_ts && !priv->samples_out) { + guint64 ts, distance; + + ts = gst_adapter_prev_timestamp (priv->adapter, &distance); + g_assert (distance % ctx->state.bpf == 0); + distance /= ctx->state.bpf; + GST_LOG_OBJECT (enc, "%" G_GUINT64_FORMAT " samples past prev_ts %" + GST_TIME_FORMAT, distance, GST_TIME_ARGS (ts)); + GST_LOG_OBJECT (enc, "%" G_GUINT64_FORMAT " samples past base_ts %" + GST_TIME_FORMAT, priv->samples, GST_TIME_ARGS (priv->base_ts)); + /* when draining adapter might be empty and no ts to offer */ + if (GST_CLOCK_TIME_IS_VALID (ts) && ts != priv->base_ts) { + GstClockTimeDiff diff; + GstClockTime old_ts, next_ts; + + /* passed into another buffer; + * mild check for discontinuity and only mark if so */ + next_ts = ts + + gst_util_uint64_scale (distance, GST_SECOND, ctx->state.rate); + old_ts = priv->base_ts + + gst_util_uint64_scale (priv->samples, GST_SECOND, ctx->state.rate); + diff = GST_CLOCK_DIFF (next_ts, old_ts); + GST_LOG_OBJECT (enc, "ts diff %d ms", (gint) (diff / GST_MSECOND)); + /* only mark discontinuity if beyond tolerance */ + if (G_UNLIKELY (diff < -enc->tolerance || diff > enc->tolerance)) { + GST_DEBUG_OBJECT (enc, "marked discont"); + priv->discont = TRUE; + } + GST_LOG_OBJECT (enc, "new upstream ts %" GST_TIME_FORMAT + " at distance %" G_GUINT64_FORMAT, GST_TIME_ARGS (ts), distance); + /* re-sync to upstream ts */ + priv->base_ts = ts; + priv->samples = distance; + } + } + /* advance sample view */ + if (G_UNLIKELY (samples * ctx->state.bpf > priv->offset)) { + if (G_LIKELY (!priv->force)) { + /* no way we can let this pass */ + g_assert_not_reached (); + /* really no way */ + goto overflow; + } else { + priv->offset = 0; + if (samples * ctx->state.bpf >= gst_adapter_available (priv->adapter)) + gst_adapter_clear (priv->adapter); + else + gst_adapter_flush (priv->adapter, samples * ctx->state.bpf); + } + } else { + gst_adapter_flush (priv->adapter, samples * ctx->state.bpf); + priv->offset -= samples * ctx->state.bpf; + /* avoid subsequent stray prev_ts */ + if (G_UNLIKELY (gst_adapter_available (priv->adapter) == 0)) + gst_adapter_clear (priv->adapter); + } + if (G_LIKELY (buf)) { + priv->samples_out += samples; + samples = 0; + } + /* otherwise retain count of to-be-discarded samples */ + } + + /* collect output */ + if (G_LIKELY (buf)) + gst_adapter_push (enc->priv->adapter_out, buf); + + av = gst_adapter_available (priv->adapter_out); + if (av) { + GST_LOG_OBJECT (enc, "collecting all %d bytes for output", av); + buf = gst_adapter_take_buffer (priv->adapter_out, av); + + /* decorate */ + gst_buffer_set_caps (buf, GST_PAD_CAPS (enc->srcpad)); + if (G_LIKELY (GST_CLOCK_TIME_IS_VALID (priv->base_ts))) { + /* FIXME ? lookahead could lead to weird ts and duration ? + * (particularly if not in perfect mode) */ + /* mind sample rounding and produce perfect output */ + GST_BUFFER_TIMESTAMP (buf) = priv->base_ts + + gst_util_uint64_scale (priv->samples - ctx->lookahead, GST_SECOND, + ctx->state.rate); + GST_DEBUG_OBJECT (enc, "out samples %d", (gint) priv->samples_out); + if (G_LIKELY (priv->samples_out > 0)) { + priv->samples += priv->samples_out; + priv->samples_out = 0; + GST_BUFFER_DURATION (buf) = priv->base_ts + + gst_util_uint64_scale (priv->samples - ctx->lookahead, GST_SECOND, + ctx->state.rate) - GST_BUFFER_TIMESTAMP (buf); + priv->last_duration = GST_BUFFER_DURATION (buf); + } else { + /* duration forecast in case of handling remainder; + * the last one is probably like the previous one ... */ + GST_BUFFER_DURATION (buf) = priv->last_duration; + } + if (priv->base_gp >= 0) { + /* pamper oggmux */ + /* FIXME: in longer run, muxer should take care of this ... */ + /* offset_end = granulepos for ogg muxer */ + GST_BUFFER_OFFSET_END (buf) = priv->base_gp + priv->samples - + enc->ctx->lookahead; + /* offset = timestamp corresponding to granulepos for ogg muxer */ + GST_BUFFER_OFFSET (buf) = + GST_FRAMES_TO_CLOCK_TIME (GST_BUFFER_OFFSET_END (buf), + ctx->state.rate); + } else { + GST_BUFFER_OFFSET (buf) = priv->bytes_out; + GST_BUFFER_OFFSET_END (buf) = priv->bytes_out + GST_BUFFER_SIZE (buf); + } + } + + if (G_UNLIKELY (priv->discont)) { + GST_LOG_OBJECT (enc, "marking discont"); + GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT); + priv->discont = FALSE; + } + // TODO return value ? + if (klass->pre_push) + klass->pre_push (enc, buf); + + GST_LOG_OBJECT (enc, "pushing buffer of size %d with ts %" GST_TIME_FORMAT + ", duration %" GST_TIME_FORMAT, GST_BUFFER_SIZE (buf), + GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)), + GST_TIME_ARGS (GST_BUFFER_DURATION (buf))); + + priv->bytes_out += GST_BUFFER_SIZE (buf); + + ret = gst_pad_push (enc->srcpad, buf); + GST_LOG_OBJECT (enc, "buffer pushed: %s", gst_flow_get_name (ret)); + } + + /* account for discarded */ + priv->samples += samples; + + return ret; + + /* ERRORS */ +overflow: + { + GST_ELEMENT_ERROR (enc, STREAM, ENCODE, + ("received more encoded samples %d than provided %d", + samples, priv->offset / ctx->state.bpf), (NULL)); + if (buf) + gst_buffer_unref (buf); + return GST_FLOW_ERROR; + } +} + + /* adapter tracking idea: + * - start of adapter corresponds with what has already been encoded + * (i.e. really returned by encoder subclass) + * - start + offset is what needs to be fed to subclass next */ +static GstFlowReturn +gst_base_audio_encoder_push_buffers (GstBaseAudioEncoder * enc, gboolean force) +{ + GstBaseAudioEncoderClass *klass; + GstBaseAudioEncoderPrivate *priv; + GstBaseAudioEncoderContext *ctx; + gint av, need; + GstBuffer *buf; + GstFlowReturn ret = GST_FLOW_OK; + + klass = GST_BASE_AUDIO_ENCODER_GET_CLASS (enc); + + g_return_val_if_fail (klass->handle_frame != NULL, GST_FLOW_ERROR); + + priv = enc->priv; + ctx = enc->ctx; + + /* ensure clear start */ + gst_adapter_clear (priv->adapter_out); + priv->samples_out = 0; + + while (ret == GST_FLOW_OK) { + + buf = NULL; + av = gst_adapter_available (priv->adapter); + + g_assert (priv->offset <= av); + av -= priv->offset; + + need = ctx->frame_samples > 0 ? ctx->frame_samples * ctx->state.bpf : av; + GST_LOG_OBJECT (enc, "available: %d, needed: %d, force: %d", + av, need, force); + + if ((need > av) || !av) { + if (G_UNLIKELY (force)) { + priv->force = TRUE; + need = av; + } else { + break; + } + } else { + priv->force = FALSE; + } + + /* if we have some extra metadata, + * provide for integer multiple of frames to allow for better granularity + * of processing */ + if (ctx->frame_samples > 0 && need) { + if (ctx->frame_max > 1) + need = need * MIN ((av / need), ctx->frame_max); + else if (ctx->frame_max == 0) + need = need * (av / need); + } + + if (need) { + buf = gst_buffer_new (); + GST_BUFFER_DATA (buf) = (guint8 *) + gst_adapter_peek (priv->adapter, priv->offset + need) + priv->offset; + GST_BUFFER_SIZE (buf) = need; + } + + GST_LOG_OBJECT (enc, "providing subclass with %d bytes at offset %d", + need, priv->offset); + + /* mark this already as consumed, + * which it should be when subclass gives us data in exchange for samples */ + priv->offset += need; + priv->samples_in += need / ctx->state.bpf; + + priv->got_data = FALSE; + ret = klass->handle_frame (enc, buf); + + if (G_LIKELY (buf)) + gst_buffer_unref (buf); + + /* no data to feed, no leftover provided, then bail out */ + if (G_UNLIKELY (!buf && !priv->got_data)) { + priv->drained = TRUE; + GST_LOG_OBJECT (enc, "no more data drained from subclass"); + break; + } + } + + if (gst_adapter_available (priv->adapter_out)) + gst_base_audio_encoder_finish_frame (enc, NULL, 0); + + return ret; +} + +static GstFlowReturn +gst_base_audio_encoder_drain (GstBaseAudioEncoder * enc) +{ + if (enc->priv->drained) + return GST_FLOW_OK; + else + return gst_base_audio_encoder_push_buffers (enc, TRUE); +} + +static GstFlowReturn +gst_base_audio_encoder_chain (GstPad * pad, GstBuffer * buffer) +{ + GstBaseAudioEncoderClass *bclass; + GstBaseAudioEncoder *enc; + GstBaseAudioEncoderPrivate *priv; + GstBaseAudioEncoderContext *ctx; + GstFlowReturn ret = GST_FLOW_OK; + gboolean discont; + + enc = GST_BASE_AUDIO_ENCODER (GST_OBJECT_PARENT (pad)); + bclass = GST_BASE_AUDIO_ENCODER_GET_CLASS (enc); + + priv = enc->priv; + ctx = enc->ctx; + + /* should know what is coming by now */ + if (!ctx->state.bpf) + goto not_negotiated; + + GST_LOG_OBJECT (enc, + "received buffer of size %d with ts %" GST_TIME_FORMAT + ", duration %" GST_TIME_FORMAT, GST_BUFFER_SIZE (buffer), + GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)), + GST_TIME_ARGS (GST_BUFFER_DURATION (buffer))); + + /* input shoud be whole number of sample frames */ + if (GST_BUFFER_SIZE (buffer) % ctx->state.bpf) + goto wrong_buffer; + +#ifndef GST_DISABLE_GST_DEBUG + { + GstClockTime duration; + GstClockTimeDiff diff; + + /* verify buffer duration */ + duration = gst_util_uint64_scale (GST_BUFFER_SIZE (buffer), GST_SECOND, + ctx->state.rate * ctx->state.bpf); + diff = GST_CLOCK_DIFF (duration, GST_BUFFER_DURATION (buffer)); + if (GST_BUFFER_DURATION (buffer) != GST_CLOCK_TIME_NONE && + (diff > GST_SECOND / ctx->state.rate / 2 || + diff < -GST_SECOND / ctx->state.rate / 2)) { + GST_DEBUG_OBJECT (enc, "incoming buffer had incorrect duration %" + GST_TIME_FORMAT ", expected duration %" GST_TIME_FORMAT, + GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)), + GST_TIME_ARGS (duration)); + } + } +#endif + + discont = GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DISCONT); + if (G_UNLIKELY (discont)) { + GST_LOG_OBJECT (buffer, "marked discont"); + enc->priv->discont = discont; + } + + /* clip to segment */ + /* NOTE: slightly painful linking -laudio only for this one ... */ + buffer = gst_audio_buffer_clip (buffer, &enc->segment, ctx->state.rate, + ctx->state.bpf); + if (G_UNLIKELY (!buffer)) { + GST_DEBUG_OBJECT (buffer, "no data after clipping to segment"); + goto done; + } + + GST_LOG_OBJECT (enc, + "buffer after segment clipping has size %d with ts %" GST_TIME_FORMAT + ", duration %" GST_TIME_FORMAT, GST_BUFFER_SIZE (buffer), + GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)), + GST_TIME_ARGS (GST_BUFFER_DURATION (buffer))); + + if (!GST_CLOCK_TIME_IS_VALID (priv->base_ts)) { + priv->base_ts = GST_BUFFER_TIMESTAMP (buffer); + GST_DEBUG_OBJECT (enc, "new base ts %" GST_TIME_FORMAT, + GST_TIME_ARGS (priv->base_ts)); + if (enc->granule) { + priv->base_gp = + GST_CLOCK_TIME_TO_FRAMES (priv->base_ts, enc->ctx->state.rate); + GST_DEBUG_OBJECT (enc, "new base gp %" G_GINT64_FORMAT, + GST_TIME_ARGS (priv->base_gp)); + } + } + + /* check for continuity; + * checked elsewhere in non-perfect case */ + if (enc->perfect_ts) { + GstClockTimeDiff diff = 0; + GstClockTime next_ts = 0; + + if (GST_BUFFER_TIMESTAMP_IS_VALID (buffer) && + GST_CLOCK_TIME_IS_VALID (priv->base_ts)) { + guint64 samples; + + samples = priv->samples + + gst_adapter_available (priv->adapter) / ctx->state.bpf; + next_ts = priv->base_ts + + gst_util_uint64_scale (samples, GST_SECOND, ctx->state.rate); + GST_LOG_OBJECT (enc, "buffer is %" G_GUINT64_FORMAT + " samples past base_ts %" GST_TIME_FORMAT + ", expected ts %" GST_TIME_FORMAT, samples, + GST_TIME_ARGS (priv->base_ts), GST_TIME_ARGS (next_ts)); + diff = GST_CLOCK_DIFF (next_ts, GST_BUFFER_TIMESTAMP (buffer)); + GST_LOG_OBJECT (enc, "ts diff %d ms", (gint) (diff / GST_MSECOND)); + /* if within tolerance, + * discard buffer ts and carry on producing perfect stream, + * otherwise clip or resync to ts */ + if (G_UNLIKELY (diff < -enc->tolerance || diff > enc->tolerance)) { + GST_DEBUG_OBJECT (enc, "marked discont"); + discont = TRUE; + } + } + + /* do some fancy tweaking in hard resync case */ + if (discont && enc->hard_resync) { + if (diff < 0) { + guint64 diff_bytes; + + GST_WARNING_OBJECT (enc, "Buffer is older than expected ts %" + GST_TIME_FORMAT ". Clipping buffer", GST_TIME_ARGS (next_ts)); + + diff_bytes = + GST_CLOCK_TIME_TO_FRAMES (-diff, ctx->state.rate) * ctx->state.bpf; + if (diff_bytes >= GST_BUFFER_SIZE (buffer)) { + gst_buffer_unref (buffer); + goto done; + } + buffer = gst_buffer_make_metadata_writable (buffer); + GST_BUFFER_DATA (buffer) += diff_bytes; + GST_BUFFER_SIZE (buffer) -= diff_bytes; + + GST_BUFFER_TIMESTAMP (buffer) += diff; + /* care even less about duration after this */ + } else { + /* drain stuff prior to resync */ + gst_base_audio_encoder_drain (enc); + } + } + /* now re-sync ts */ + priv->base_ts += diff; + if (priv->base_gp >= 0) + priv->base_gp = + GST_CLOCK_TIME_TO_FRAMES (priv->base_ts, enc->ctx->state.rate); + priv->discont |= discont; + } + + gst_adapter_push (enc->priv->adapter, buffer); + /* new stuff, so we can push subclass again */ + enc->priv->drained = FALSE; + + ret = gst_base_audio_encoder_push_buffers (enc, FALSE); + +done: + GST_LOG_OBJECT (enc, "chain leaving"); + return ret; + + /* ERRORS */ +not_negotiated: + { + GST_ELEMENT_ERROR (enc, CORE, NEGOTIATION, (NULL), + ("encoder not initialized")); + gst_buffer_unref (buffer); + return GST_FLOW_NOT_NEGOTIATED; + } +wrong_buffer: + { + GST_ELEMENT_ERROR (enc, STREAM, ENCODE, (NULL), + ("buffer size %d not a multiple of %d", GST_BUFFER_SIZE (buffer), + ctx->state.bpf)); + gst_buffer_unref (buffer); + return GST_FLOW_ERROR; + } +} + +#define CHECK_VALUE(res, var, val) \ + if (!res) \ + goto refuse_caps; \ + if (var != val) \ + changed = TRUE; \ + var = val; + +static gboolean +gst_base_audio_encoder_sink_setcaps (GstPad * pad, GstCaps * caps) +{ + GstBaseAudioEncoder *enc; + GstBaseAudioEncoderClass *klass; + GstBaseAudioEncoderContext *ctx; + GstAudioState *state; + gboolean res = TRUE, changed = FALSE; + GstStructure *s; + gboolean vb; + gint vi; + + enc = GST_BASE_AUDIO_ENCODER (GST_PAD_PARENT (pad)); + klass = GST_BASE_AUDIO_ENCODER_GET_CLASS (enc); + + /* subclass must do something here ... */ + g_return_val_if_fail (klass->set_format != NULL, FALSE); + + ctx = enc->ctx; + state = &ctx->state; + + GST_DEBUG_OBJECT (enc, "caps: %" GST_PTR_FORMAT, caps); + + if (!gst_caps_is_fixed (caps)) + goto refuse_caps; + + s = gst_caps_get_structure (caps, 0); + /* parse caps here to save subclass the trouble */ + if (gst_structure_has_name (s, "audio/x-raw-int")) + state->xint = TRUE; + else if (gst_structure_has_name (s, "audio/x-raw-float")) + state->xint = FALSE; + else + goto refuse_caps; + + res = gst_structure_get_int (s, "rate", &vi); + CHECK_VALUE (res, state->rate, vi); + res &= gst_structure_get_int (s, "channels", &vi); + CHECK_VALUE (res, state->channels, vi); + res &= gst_structure_get_int (s, "width", &vi); + CHECK_VALUE (res, state->width, vi); + res &= (!state->xint || gst_structure_get_int (s, "depth", &vi)); + CHECK_VALUE (res, state->depth, vi); + res &= gst_structure_get_int (s, "endianness", &vi); + CHECK_VALUE (res, state->endian, vi); + res &= (!state->xint || gst_structure_get_boolean (s, "signed", &vb)); + CHECK_VALUE (res, state->sign, vb); + + state->bpf = (state->width / 8) * state->channels; + GST_LOG_OBJECT (enc, "bpf: %d", state->bpf); + if (!state->bpf) + goto refuse_caps; + + g_free (state->channel_pos); + state->channel_pos = gst_audio_get_channel_positions (s); + + if (changed) { + GstClockTime old_min_latency; + GstClockTime old_max_latency; + + /* drain any pending old data stuff */ + gst_base_audio_encoder_drain (enc); + + /* context defaults */ + enc->ctx->frame_samples = 0; + enc->ctx->frame_max = 0; + enc->ctx->lookahead = 0; + + /* element might report latency */ + old_min_latency = ctx->min_latency; + old_max_latency = ctx->max_latency; + + if (klass->set_format) + res = klass->set_format (enc, state); + + /* notify if new latency */ + if ((ctx->min_latency > 0 && ctx->min_latency != old_min_latency) || + (ctx->max_latency > 0 && ctx->max_latency != old_max_latency)) { + /* post latency message on the bus */ + gst_element_post_message (GST_ELEMENT (enc), + gst_message_new_latency (GST_OBJECT (enc))); + GST_OBJECT_LOCK (enc); + enc->priv->min_latency = ctx->min_latency; + enc->priv->max_latency = ctx->max_latency; + GST_OBJECT_UNLOCK (enc); + } + } else { + GST_DEBUG_OBJECT (enc, "new audio format identical to configured format"); + } + + return res; + + /* ERRORS */ +refuse_caps: + { + GST_WARNING_OBJECT (enc, "rejected caps %" GST_PTR_FORMAT, caps); + return res; + } +} + + +/** gst_base_audio_encoder_proxy_getcaps: + * @enc: a #GstBaseAudioEncoder + * @caps: initial + * + * Returns caps that express @caps (or sink template caps if @caps == NULL) + * restricted to channel/rate combinations supported by downstream elements + * (e.g. muxers). + * + * Returns: a #GstCaps owned by caller + */ +GstCaps * +gst_base_audio_encoder_proxy_getcaps (GstBaseAudioEncoder * enc, GstCaps * caps) +{ + const GstCaps *templ_caps; + GstCaps *allowed = NULL; + GstCaps *fcaps, *filter_caps; + gint i, j; + + /* we want to be able to communicate to upstream elements like audioconvert + * and audioresample any rate/channel restrictions downstream (e.g. muxer + * only accepting certain sample rates) */ + templ_caps = caps ? caps : gst_pad_get_pad_template_caps (enc->sinkpad); + allowed = gst_pad_get_allowed_caps (enc->srcpad); + if (!allowed || gst_caps_is_empty (allowed) || gst_caps_is_any (allowed)) { + fcaps = gst_caps_copy (templ_caps); + goto done; + } + + GST_LOG_OBJECT (enc, "template caps %" GST_PTR_FORMAT, templ_caps); + GST_LOG_OBJECT (enc, "allowed caps %" GST_PTR_FORMAT, allowed); + + filter_caps = gst_caps_new_empty (); + + for (i = 0; i < gst_caps_get_size (templ_caps); i++) { + GQuark q_name; + + q_name = gst_structure_get_name_id (gst_caps_get_structure (templ_caps, i)); + + /* pick rate + channel fields from allowed caps */ + for (j = 0; j < gst_caps_get_size (allowed); j++) { + const GstStructure *allowed_s = gst_caps_get_structure (allowed, j); + const GValue *val; + GstStructure *s; + + s = gst_structure_id_empty_new (q_name); + if ((val = gst_structure_get_value (allowed_s, "rate"))) + gst_structure_set_value (s, "rate", val); + if ((val = gst_structure_get_value (allowed_s, "channels"))) + gst_structure_set_value (s, "channels", val); + + gst_caps_merge_structure (filter_caps, s); + } + } + + fcaps = gst_caps_intersect (filter_caps, templ_caps); + gst_caps_unref (filter_caps); + +done: + gst_caps_replace (&allowed, NULL); + + GST_LOG_OBJECT (enc, "proxy caps %" GST_PTR_FORMAT, fcaps); + + return fcaps; +} + +static GstCaps * +gst_base_audio_encoder_sink_getcaps (GstPad * pad) +{ + GstBaseAudioEncoder *enc; + GstBaseAudioEncoderClass *klass; + GstCaps *caps; + + enc = GST_BASE_AUDIO_ENCODER (gst_pad_get_parent (pad)); + klass = GST_BASE_AUDIO_ENCODER_GET_CLASS (enc); + g_assert (pad == enc->sinkpad); + + if (klass->getcaps) + caps = klass->getcaps (enc); + else + caps = gst_base_audio_encoder_proxy_getcaps (enc, NULL); + gst_object_unref (enc); + + GST_LOG_OBJECT (enc, "returning caps %" GST_PTR_FORMAT, caps); + + return caps; +} + +static gboolean +gst_base_audio_encoder_sink_eventfunc (GstBaseAudioEncoder * enc, + GstEvent * event) +{ + GstBaseAudioEncoderClass *klass; + gboolean handled = FALSE; + + klass = GST_BASE_AUDIO_ENCODER_GET_CLASS (enc); + + switch (GST_EVENT_TYPE (event)) { + case GST_EVENT_NEWSEGMENT: + { + GstFormat format; + gdouble rate, arate; + gint64 start, stop, time; + gboolean update; + + gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format, + &start, &stop, &time); + + if (format == GST_FORMAT_TIME) { + GST_DEBUG_OBJECT (enc, "received TIME NEW_SEGMENT %" GST_TIME_FORMAT + " -- %" GST_TIME_FORMAT ", time %" GST_TIME_FORMAT + ", accum %" GST_TIME_FORMAT, + GST_TIME_ARGS (enc->segment.start), + GST_TIME_ARGS (enc->segment.stop), + GST_TIME_ARGS (enc->segment.time), + GST_TIME_ARGS (enc->segment.accum)); + } else { + GST_DEBUG_OBJECT (enc, "received NEW_SEGMENT %" G_GINT64_FORMAT + " -- %" G_GINT64_FORMAT ", time %" G_GINT64_FORMAT + ", accum %" G_GINT64_FORMAT, + enc->segment.start, enc->segment.stop, + enc->segment.time, enc->segment.accum); + GST_DEBUG_OBJECT (enc, "unsupported format; ignoring"); + break; + } + + /* finish current segment */ + gst_base_audio_encoder_drain (enc); + /* reset partially for new segment */ + gst_base_audio_encoder_reset (enc, FALSE); + /* and follow along with segment */ + gst_segment_set_newsegment_full (&enc->segment, update, rate, arate, + format, start, stop, time); + break; + } + + case GST_EVENT_FLUSH_START: + break; + + case GST_EVENT_FLUSH_STOP: + /* discard any pending stuff */ + // TODO route through drain ? + if (!enc->priv->drained && klass->flush) + klass->flush (enc); + /* and get (re)set for the sequel */ + gst_base_audio_encoder_reset (enc, FALSE); + break; + + case GST_EVENT_EOS: + gst_base_audio_encoder_drain (enc); + break; + + default: + break; + } + + return handled; +} + +static gboolean +gst_base_audio_encoder_sink_event (GstPad * pad, GstEvent * event) +{ + GstBaseAudioEncoder *enc; + GstBaseAudioEncoderClass *klass; + gboolean handled = FALSE; + gboolean ret = TRUE; + + enc = GST_BASE_AUDIO_ENCODER (gst_pad_get_parent (pad)); + klass = GST_BASE_AUDIO_ENCODER_GET_CLASS (enc); + + GST_DEBUG_OBJECT (enc, "received event %d, %s", GST_EVENT_TYPE (event), + GST_EVENT_TYPE_NAME (event)); + + if (klass->event) + handled = klass->event (enc, event); + + if (!handled) + handled = gst_base_audio_encoder_sink_eventfunc (enc, event); + + if (!handled) + ret = gst_pad_event_default (pad, event); + + GST_DEBUG_OBJECT (enc, "event handled"); + + gst_object_unref (enc); + return ret; +} + +static gboolean +gst_base_audio_encoder_convert_sink (GstPad * pad, GstFormat src_format, + gint64 src_value, GstFormat * dest_format, gint64 * dest_value) +{ + GstBaseAudioEncoder *enc; + gboolean res = FALSE; + guint scale = 1; + gint bytes_per_sample, rate, byterate; + + enc = GST_BASE_AUDIO_ENCODER (gst_pad_get_parent (pad)); + + bytes_per_sample = enc->ctx->state.bpf; + rate = enc->ctx->state.rate; + byterate = bytes_per_sample * rate; + + if (G_UNLIKELY (bytes_per_sample == 0 || rate == 0)) { + GST_DEBUG_OBJECT (enc, "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: + gst_object_unref (enc); + return res; +} + +static gboolean +gst_base_audio_encoder_sink_query (GstPad * pad, GstQuery * query) +{ + gboolean res = TRUE; + + switch (GST_QUERY_TYPE (query)) { + case GST_QUERY_FORMATS: + { + gst_query_set_formats (query, 3, + GST_FORMAT_TIME, GST_FORMAT_BYTES, GST_FORMAT_DEFAULT); + res = TRUE; + break; + } + case GST_QUERY_CONVERT: + { + GstFormat src_fmt, dest_fmt; + gint64 src_val, dest_val; + + gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val); + if (!(res = + gst_base_audio_encoder_convert_sink (pad, src_fmt, src_val, + &dest_fmt, &dest_val))) + goto error; + gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val); + break; + } + default: + res = gst_pad_query_default (pad, query); + break; + } + +error: + return res; +} + +static gboolean +gst_base_audio_encoder_convert_src (GstPad * pad, GstFormat src_format, + gint64 src_value, GstFormat * dest_format, gint64 * dest_value) +{ + GstBaseAudioEncoder *enc; + gboolean res = FALSE; + gint64 avg; + + enc = GST_BASE_AUDIO_ENCODER (gst_pad_get_parent (pad)); + + if (enc->priv->samples_in == 0 || + enc->priv->bytes_out == 0 || enc->ctx->state.rate == 0) { + GST_DEBUG_OBJECT (enc, "not enough metadata yet to convert"); + goto exit; + } + + avg = (enc->priv->bytes_out * enc->ctx->state.rate) / (enc->priv->samples_in); + + switch (src_format) { + case GST_FORMAT_BYTES: + switch (*dest_format) { + case GST_FORMAT_TIME: + *dest_value = gst_util_uint64_scale_int (src_value, GST_SECOND, avg); + res = TRUE; + break; + default: + res = FALSE; + } + break; + case GST_FORMAT_TIME: + switch (*dest_format) { + case GST_FORMAT_BYTES: + *dest_value = gst_util_uint64_scale_int (src_value, avg, GST_SECOND); + res = TRUE; + break; + default: + res = FALSE; + } + break; + default: + res = FALSE; + } + +exit: + gst_object_unref (enc); + return res; +} + +static const GstQueryType * +gst_base_audio_encoder_get_query_types (GstPad * pad) +{ + static const GstQueryType gst_base_audio_encoder_src_query_types[] = { + GST_QUERY_POSITION, + GST_QUERY_DURATION, + GST_QUERY_CONVERT, + GST_QUERY_LATENCY, + 0 + }; + + return gst_base_audio_encoder_src_query_types; +} + +/* FIXME ? are any of these queries (other than latency) an encoder's business + * also, the conversion stuff might seem to make sense, but seems to not mind + * segment stuff etc at all + * Supposedly that's backward compatibility ... */ +static gboolean +gst_base_audio_encoder_src_query (GstPad * pad, GstQuery * query) +{ + GstBaseAudioEncoder *enc; + GstPad *peerpad; + gboolean res = FALSE; + + enc = GST_BASE_AUDIO_ENCODER (GST_PAD_PARENT (pad)); + peerpad = gst_pad_get_peer (GST_PAD (enc->sinkpad)); + + GST_LOG_OBJECT (enc, "handling query: %" GST_PTR_FORMAT, query); + + switch (GST_QUERY_TYPE (query)) { + case GST_QUERY_POSITION: + { + GstFormat fmt, req_fmt; + gint64 pos, val; + + if ((res = gst_pad_peer_query (pad, query))) { + GST_LOG_OBJECT (enc, "returning peer response"); + break; + } + + if (!peerpad) { + GST_LOG_OBJECT (enc, "no peer"); + break; + } + + gst_query_parse_position (query, &req_fmt, NULL); + fmt = GST_FORMAT_TIME; + if (!(res = gst_pad_query_position (peerpad, &fmt, &pos))) + break; + + if ((res = gst_pad_query_convert (peerpad, fmt, pos, &req_fmt, &val))) { + gst_query_set_position (query, req_fmt, val); + } + break; + } + case GST_QUERY_DURATION: + { + GstFormat fmt, req_fmt; + gint64 dur, val; + + if ((res = gst_pad_peer_query (pad, query))) { + GST_LOG_OBJECT (enc, "returning peer response"); + break; + } + + if (!peerpad) { + GST_LOG_OBJECT (enc, "no peer"); + break; + } + + gst_query_parse_duration (query, &req_fmt, NULL); + fmt = GST_FORMAT_TIME; + if (!(res = gst_pad_query_duration (peerpad, &fmt, &dur))) + break; + + if ((res = gst_pad_query_convert (peerpad, fmt, dur, &req_fmt, &val))) { + gst_query_set_duration (query, req_fmt, val); + } + break; + } + case GST_QUERY_FORMATS: + { + gst_query_set_formats (query, 2, GST_FORMAT_TIME, GST_FORMAT_BYTES); + res = TRUE; + break; + } + case GST_QUERY_CONVERT: + { + GstFormat src_fmt, dest_fmt; + gint64 src_val, dest_val; + + gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val); + if (!(res = gst_base_audio_encoder_convert_src (pad, src_fmt, src_val, + &dest_fmt, &dest_val))) + break; + gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val); + break; + } + case GST_QUERY_LATENCY: + { + if ((res = gst_pad_peer_query (pad, query))) { + gboolean live; + GstClockTime min_latency, max_latency; + + gst_query_parse_latency (query, &live, &min_latency, &max_latency); + GST_DEBUG_OBJECT (enc, "Peer latency: live %d, min %" + GST_TIME_FORMAT " max %" GST_TIME_FORMAT, live, + GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency)); + + GST_OBJECT_LOCK (enc); + /* add our latency */ + if (min_latency != -1) + min_latency += enc->priv->min_latency; + if (max_latency != -1) + max_latency += enc->priv->max_latency; + GST_OBJECT_UNLOCK (enc); + + gst_query_set_latency (query, live, min_latency, max_latency); + } + } + default: + res = gst_pad_query_default (pad, query); + break; + } + + gst_object_unref (peerpad); + return res; +} + +static void +gst_base_audio_encoder_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec) +{ + GstBaseAudioEncoder *enc; + + enc = GST_BASE_AUDIO_ENCODER (object); + + switch (prop_id) { + case PROP_PERFECT_TS: + if (enc->granule && !g_value_get_boolean (value)) + GST_WARNING_OBJECT (enc, "perfect-ts can not be set FALSE"); + else + enc->perfect_ts = g_value_get_boolean (value); + break; + case PROP_HARD_RESYNC: + enc->hard_resync = g_value_get_boolean (value); + break; + case PROP_TOLERANCE: + enc->tolerance = g_value_get_int64 (value); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static void +gst_base_audio_encoder_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec) +{ + GstBaseAudioEncoder *enc; + + enc = GST_BASE_AUDIO_ENCODER (object); + + switch (prop_id) { + case PROP_PERFECT_TS: + g_value_set_boolean (value, enc->perfect_ts); + break; + case PROP_GRANULE: + g_value_set_boolean (value, enc->granule); + break; + case PROP_HARD_RESYNC: + g_value_set_boolean (value, enc->hard_resync); + break; + case PROP_TOLERANCE: + g_value_set_int64 (value, enc->tolerance); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static gboolean +gst_base_audio_encoder_activate (GstBaseAudioEncoder * enc, gboolean active) +{ + GstBaseAudioEncoderClass *klass; + gboolean result = FALSE; + + klass = GST_BASE_AUDIO_ENCODER_GET_CLASS (enc); + + g_return_val_if_fail (!enc->granule || enc->perfect_ts, FALSE); + + GST_DEBUG_OBJECT (enc, "activate %d", active); + + if (active) { + if (!enc->priv->active && klass->start) + result = klass->start (enc); + } else { + /* We must make sure streaming has finished before resetting things + * and calling the ::stop vfunc */ + GST_PAD_STREAM_LOCK (enc->sinkpad); + GST_PAD_STREAM_UNLOCK (enc->sinkpad); + + if (enc->priv->active && klass->stop) + result = klass->stop (enc); + + /* clean up */ + gst_base_audio_encoder_reset (enc, TRUE); + } + GST_DEBUG_OBJECT (enc, "activate return: %d", result); + return result; +} + + +static gboolean +gst_base_audio_encoder_sink_activate_push (GstPad * pad, gboolean active) +{ + gboolean result = TRUE; + GstBaseAudioEncoder *enc; + + enc = GST_BASE_AUDIO_ENCODER (gst_pad_get_parent (pad)); + + GST_DEBUG_OBJECT (enc, "sink activate push %d", active); + + result = gst_base_audio_encoder_activate (enc, active); + + if (result) + enc->priv->active = active; + + GST_DEBUG_OBJECT (enc, "sink activate push return: %d", result); + + gst_object_unref (enc); + return result; +} diff --git a/gst-libs/gst/audio/gstbaseaudioencoder.h b/gst-libs/gst/audio/gstbaseaudioencoder.h new file mode 100644 index 0000000000..27c1ba0710 --- /dev/null +++ b/gst-libs/gst/audio/gstbaseaudioencoder.h @@ -0,0 +1,251 @@ +/* GStreamer + * Copyright (C) 2011 Mark Nauwelaerts . + * Copyright (C) 2011 Nokia Corporation. All rights reserved. + * Contact: Stefan Kost + * + * 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__ + +#include +#include +#include + +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) + +/** + * GST_BASE_AUDIO_ENCODER_FLOW_DROPPED: + * + * A #GstFlowReturn that can be returned from parse_frame to + * indicate that no output buffer was generated, or from pre_push_buffer to + * to forego pushing buffer. + * + * Since: 0.10.x + */ +#define GST_BASE_AUDIO_ENCODER_FLOW_DROPPED GST_FLOW_CUSTOM_SUCCESS + + +typedef struct _GstBaseAudioEncoder GstBaseAudioEncoder; +typedef struct _GstBaseAudioEncoderClass GstBaseAudioEncoderClass; + +typedef struct _GstBaseAudioEncoderPrivate GstBaseAudioEncoderPrivate; +typedef struct _GstBaseAudioEncoderContext GstBaseAudioEncoderContext; + +/** + * GstAudioState: + * @xint: 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: rate of sample data (if integer) + * @endian: endianness of sample data + * @bpf: bytes per audio frame + */ +typedef struct _GstAudioState { + gboolean xint; + gint rate; + gint channels; + gint width; + gint depth; + gboolean sign; + gint endian; + GstAudioChannelPosition *channel_pos; + + gint bpf; +} GstAudioState; + +/** + * GstBaseAudioEncoderContext: + * @state: a #GstAudioState describing input audio format + * @frame_samples: number of samples (per channel) subclass needs to be handed, + * or will be handed all available if 0. + * @frame_max: max number of frames of size @frame_bytes accepted at once + * (assumed minimally 1) + * @latency: latency of element; should only be changed during configure + * @lookahead: encoder lookahead (in units of input rate samples) + * + * Transparent #GstBaseAudioEncoderContext data structure. + */ +struct _GstBaseAudioEncoderContext { + /* input */ + GstAudioState state; + + /* output */ + gint frame_samples; + gint frame_max; + GstClockTime min_latency; + GstClockTime max_latency; + gint lookahead; +}; + +/** + * GstBaseAudioEncoder: + * @element: the parent element. + * + * The opaque #GstBaseAudioEncoder data structure. + */ +struct _GstBaseAudioEncoder { + GstElement element; + + /*< protected >*/ + /* source and sink pads */ + GstPad *sinkpad; + GstPad *srcpad; + + /* 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. + * @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__ */