diff options
author | Vincent Penquerc'h <vincent.penquerch@collabora.co.uk> | 2011-11-16 16:56:43 +0000 |
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committer | Vincent Penquerc'h <vincent.penquerch@collabora.co.uk> | 2011-11-16 17:45:00 +0000 |
commit | f978a60f38ad536efb688df574cf2fadc4d96db2 (patch) | |
tree | 429062053de1c44a8fc104204519d83046b5dc38 /ext/opus/gstopusenc.c | |
parent | 545b87e14c5d33c8186f7206a669a646eb16cc15 (diff) | |
download | gstreamer-plugins-base-f978a60f38ad536efb688df574cf2fadc4d96db2.tar.gz |
opus: port to base audio encoder/decoder
Diffstat (limited to 'ext/opus/gstopusenc.c')
-rw-r--r-- | ext/opus/gstopusenc.c | 901 |
1 files changed, 293 insertions, 608 deletions
diff --git a/ext/opus/gstopusenc.c b/ext/opus/gstopusenc.c index 8d40cdf81..4be63cb88 100644 --- a/ext/opus/gstopusenc.c +++ b/ext/opus/gstopusenc.c @@ -47,6 +47,7 @@ #include <gst/gsttagsetter.h> #include <gst/tag/tag.h> +#include <gst/base/gstbytewriter.h> #include <gst/audio/audio.h> #include "gstopusenc.h" @@ -125,18 +126,26 @@ enum static void gst_opus_enc_finalize (GObject * object); -static gboolean gst_opus_enc_sinkevent (GstPad * pad, GstEvent * event); -static GstFlowReturn gst_opus_enc_chain (GstPad * pad, GstBuffer * buf); +static gboolean gst_opus_enc_sink_event (GstAudioEncoder * benc, + GstEvent * event); static gboolean gst_opus_enc_setup (GstOpusEnc * enc); static void gst_opus_enc_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static void gst_opus_enc_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); -static GstStateChangeReturn gst_opus_enc_change_state (GstElement * element, - GstStateChange transition); -static GstFlowReturn gst_opus_enc_encode (GstOpusEnc * enc, gboolean flush); +static gboolean gst_opus_enc_start (GstAudioEncoder * benc); +static gboolean gst_opus_enc_stop (GstAudioEncoder * benc); +static gboolean gst_opus_enc_set_format (GstAudioEncoder * benc, + GstAudioInfo * info); +static GstFlowReturn gst_opus_enc_handle_frame (GstAudioEncoder * benc, + GstBuffer * buf); +static GstFlowReturn gst_opus_enc_pre_push (GstAudioEncoder * benc, + GstBuffer ** buffer); +static gint64 gst_opus_enc_get_latency (GstOpusEnc * enc); + +static GstFlowReturn gst_opus_enc_encode (GstOpusEnc * enc, GstBuffer * buffer); static void gst_opus_enc_setup_interfaces (GType opusenc_type) @@ -156,8 +165,8 @@ gst_opus_enc_setup_interfaces (GType opusenc_type) GST_DEBUG_CATEGORY_INIT (opusenc_debug, "opusenc", 0, "Opus encoder"); } -GST_BOILERPLATE_FULL (GstOpusEnc, gst_opus_enc, GstElement, GST_TYPE_ELEMENT, - gst_opus_enc_setup_interfaces); +GST_BOILERPLATE_FULL (GstOpusEnc, gst_opus_enc, GstAudioEncoder, + GST_TYPE_AUDIO_ENCODER, gst_opus_enc_setup_interfaces); static void gst_opus_enc_base_init (gpointer g_class) @@ -179,13 +188,22 @@ gst_opus_enc_class_init (GstOpusEncClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; + GstAudioEncoderClass *base_class; gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; + base_class = (GstAudioEncoderClass *) klass; gobject_class->set_property = gst_opus_enc_set_property; gobject_class->get_property = gst_opus_enc_get_property; + base_class->start = GST_DEBUG_FUNCPTR (gst_opus_enc_start); + base_class->stop = GST_DEBUG_FUNCPTR (gst_opus_enc_stop); + base_class->set_format = GST_DEBUG_FUNCPTR (gst_opus_enc_set_format); + base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_opus_enc_handle_frame); + base_class->pre_push = GST_DEBUG_FUNCPTR (gst_opus_enc_pre_push); + base_class->event = GST_DEBUG_FUNCPTR (gst_opus_enc_sink_event); + g_object_class_install_property (gobject_class, PROP_AUDIO, g_param_spec_boolean ("audio", "Audio or voice", "Audio or voice", DEFAULT_AUDIO, @@ -229,9 +247,6 @@ gst_opus_enc_class_init (GstOpusEncClass * klass) G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_opus_enc_finalize); - - gstelement_class->change_state = - GST_DEBUG_FUNCPTR (gst_opus_enc_change_state); } static void @@ -241,397 +256,164 @@ gst_opus_enc_finalize (GObject * object) enc = GST_OPUS_ENC (object); - g_object_unref (enc->adapter); - G_OBJECT_CLASS (parent_class)->finalize (object); } -static gboolean -gst_opus_enc_sink_setcaps (GstPad * pad, GstCaps * caps) -{ - GstOpusEnc *enc; - GstStructure *structure; - GstCaps *otherpadcaps; - - enc = GST_OPUS_ENC (GST_PAD_PARENT (pad)); - enc->setup = FALSE; - enc->frame_size = DEFAULT_FRAMESIZE; - otherpadcaps = gst_pad_get_allowed_caps (pad); - - structure = gst_caps_get_structure (caps, 0); - gst_structure_get_int (structure, "channels", &enc->n_channels); - gst_structure_get_int (structure, "rate", &enc->sample_rate); - - if (otherpadcaps) { - if (!gst_caps_is_empty (otherpadcaps)) { - GstStructure *ps = gst_caps_get_structure (otherpadcaps, 0); - gst_structure_get_int (ps, "frame-size", &enc->frame_size); - } - gst_caps_unref (otherpadcaps); - } - - GST_DEBUG_OBJECT (pad, "channels=%d rate=%d frame-size=%d", - enc->n_channels, enc->sample_rate, enc->frame_size); - switch (enc->frame_size) { - case 2: - enc->frame_samples = enc->sample_rate / 400; - break; - case 5: - enc->frame_samples = enc->sample_rate / 200; - break; - case 10: - enc->frame_samples = enc->sample_rate / 100; - break; - case 20: - enc->frame_samples = enc->sample_rate / 50; - break; - case 40: - enc->frame_samples = enc->sample_rate / 25; - break; - case 60: - enc->frame_samples = 3 * enc->sample_rate / 50; - break; - default: - GST_WARNING_OBJECT (enc, "Unsupported frame size: %d", enc->frame_size); - return FALSE; - break; - } - GST_DEBUG_OBJECT (pad, "frame_samples %d", enc->frame_samples); - - gst_opus_enc_setup (enc); - - return TRUE; -} - - -static GstCaps * -gst_opus_enc_sink_getcaps (GstPad * pad) +static void +gst_opus_enc_init (GstOpusEnc * enc, GstOpusEncClass * klass) { - GstCaps *caps = gst_caps_copy (gst_pad_get_pad_template_caps (pad)); - GstCaps *peercaps = NULL; - GstOpusEnc *enc = GST_OPUS_ENC (gst_pad_get_parent_element (pad)); - - peercaps = gst_pad_peer_get_caps (enc->srcpad); - - if (peercaps) { - if (!gst_caps_is_empty (peercaps) && !gst_caps_is_any (peercaps)) { - GstStructure *ps = gst_caps_get_structure (peercaps, 0); - GstStructure *s = gst_caps_get_structure (caps, 0); - gint rate, channels; + GstAudioEncoder *benc = GST_AUDIO_ENCODER (enc); - if (gst_structure_get_int (ps, "rate", &rate)) { - gst_structure_fixate_field_nearest_int (s, "rate", rate); - } + GST_DEBUG_OBJECT (enc, "init"); - if (gst_structure_get_int (ps, "channels", &channels)) { - gst_structure_fixate_field_nearest_int (s, "channels", channels); - } - } - gst_caps_unref (peercaps); - } + enc->n_channels = -1; + enc->sample_rate = -1; + enc->frame_samples = 0; - gst_object_unref (enc); + enc->bitrate = DEFAULT_BITRATE; + enc->bandwidth = DEFAULT_BANDWIDTH; + enc->frame_size = DEFAULT_FRAMESIZE; + enc->cbr = DEFAULT_CBR; + enc->constrained_vbr = DEFAULT_CONSTRAINED_VBR; + enc->complexity = DEFAULT_COMPLEXITY; + enc->inband_fec = DEFAULT_INBAND_FEC; + enc->dtx = DEFAULT_DTX; + enc->packet_loss_percentage = DEFAULT_PACKET_LOSS_PERCENT; - return caps; + /* arrange granulepos marking (and required perfect ts) */ + gst_audio_encoder_set_mark_granule (benc, TRUE); + gst_audio_encoder_set_perfect_timestamp (benc, TRUE); } - static gboolean -gst_opus_enc_convert_src (GstPad * pad, GstFormat src_format, gint64 src_value, - GstFormat * dest_format, gint64 * dest_value) +gst_opus_enc_start (GstAudioEncoder * benc) { - gboolean res = TRUE; - GstOpusEnc *enc; - gint64 avg; + GstOpusEnc *enc = GST_OPUS_ENC (benc); - enc = GST_OPUS_ENC (GST_PAD_PARENT (pad)); - - if (enc->samples_in == 0 || enc->bytes_out == 0 || enc->sample_rate == 0) - return FALSE; - - avg = (enc->bytes_out * enc->sample_rate) / (enc->samples_in); - - switch (src_format) { - case GST_FORMAT_BYTES: - switch (*dest_format) { - case GST_FORMAT_TIME: - *dest_value = src_value * GST_SECOND / avg; - break; - default: - res = FALSE; - } - break; - case GST_FORMAT_TIME: - switch (*dest_format) { - case GST_FORMAT_BYTES: - *dest_value = src_value * avg / GST_SECOND; - break; - default: - res = FALSE; - } - break; - default: - res = FALSE; - } - return res; + GST_DEBUG_OBJECT (enc, "start"); + enc->tags = gst_tag_list_new (); + enc->header_sent = FALSE; + return TRUE; } static gboolean -gst_opus_enc_convert_sink (GstPad * pad, GstFormat src_format, - gint64 src_value, GstFormat * dest_format, gint64 * dest_value) +gst_opus_enc_stop (GstAudioEncoder * benc) { - gboolean res = TRUE; - guint scale = 1; - gint bytes_per_sample; - GstOpusEnc *enc; + GstOpusEnc *enc = GST_OPUS_ENC (benc); - enc = GST_OPUS_ENC (GST_PAD_PARENT (pad)); - - bytes_per_sample = enc->n_channels * 2; - - switch (src_format) { - case GST_FORMAT_BYTES: - switch (*dest_format) { - case GST_FORMAT_DEFAULT: - if (bytes_per_sample == 0) - return FALSE; - *dest_value = src_value / bytes_per_sample; - break; - case GST_FORMAT_TIME: - { - gint byterate = bytes_per_sample * enc->sample_rate; - - if (byterate == 0) - return FALSE; - *dest_value = src_value * GST_SECOND / byterate; - break; - } - default: - res = FALSE; - } - break; - case GST_FORMAT_DEFAULT: - switch (*dest_format) { - case GST_FORMAT_BYTES: - *dest_value = src_value * bytes_per_sample; - break; - case GST_FORMAT_TIME: - if (enc->sample_rate == 0) - return FALSE; - *dest_value = src_value * GST_SECOND / enc->sample_rate; - 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 = src_value * scale * enc->sample_rate / GST_SECOND; - break; - default: - res = FALSE; - } - break; - default: - res = FALSE; + GST_DEBUG_OBJECT (enc, "stop"); + enc->header_sent = FALSE; + if (enc->state) { + opus_encoder_destroy (enc->state); + enc->state = NULL; } - return res; + gst_tag_list_free (enc->tags); + enc->tags = NULL; + g_slist_foreach (enc->headers, (GFunc) gst_buffer_unref, NULL); + enc->headers = NULL; + + return TRUE; } static gint64 gst_opus_enc_get_latency (GstOpusEnc * enc) { - return gst_util_uint64_scale (enc->frame_samples, GST_SECOND, + gint64 latency = gst_util_uint64_scale (enc->frame_samples, GST_SECOND, enc->sample_rate); + GST_DEBUG_OBJECT (enc, "Latency: %" GST_TIME_FORMAT, GST_TIME_ARGS (latency)); + return latency; } -static const GstQueryType * -gst_opus_enc_get_query_types (GstPad * pad) +static gint +gst_opus_enc_get_frame_samples (GstOpusEnc * enc) { - static const GstQueryType gst_opus_enc_src_query_types[] = { - GST_QUERY_POSITION, - GST_QUERY_DURATION, - GST_QUERY_CONVERT, - GST_QUERY_LATENCY, - 0 - }; - - return gst_opus_enc_src_query_types; -} - -static gboolean -gst_opus_enc_src_query (GstPad * pad, GstQuery * query) -{ - gboolean res = TRUE; - GstOpusEnc *enc; - - enc = GST_OPUS_ENC (gst_pad_get_parent (pad)); - - switch (GST_QUERY_TYPE (query)) { - case GST_QUERY_POSITION: - { - GstFormat fmt, req_fmt; - gint64 pos, val; - - gst_query_parse_position (query, &req_fmt, NULL); - if ((res = gst_pad_query_peer_position (enc->sinkpad, &req_fmt, &val))) { - gst_query_set_position (query, req_fmt, val); - break; - } - - fmt = GST_FORMAT_TIME; - if (!(res = gst_pad_query_peer_position (enc->sinkpad, &fmt, &pos))) - break; - - if ((res = - gst_pad_query_peer_convert (enc->sinkpad, fmt, pos, &req_fmt, - &val))) - gst_query_set_position (query, req_fmt, val); - + gint frame_samples = 0; + switch (enc->frame_size) { + case 2: + frame_samples = enc->sample_rate / 400; break; - } - case GST_QUERY_DURATION: - { - GstFormat fmt, req_fmt; - gint64 dur, val; - - gst_query_parse_duration (query, &req_fmt, NULL); - if ((res = gst_pad_query_peer_duration (enc->sinkpad, &req_fmt, &val))) { - gst_query_set_duration (query, req_fmt, val); - break; - } - - fmt = GST_FORMAT_TIME; - if (!(res = gst_pad_query_peer_duration (enc->sinkpad, &fmt, &dur))) - break; - - if ((res = - gst_pad_query_peer_convert (enc->sinkpad, fmt, dur, &req_fmt, - &val))) { - gst_query_set_duration (query, req_fmt, val); - } + case 5: + frame_samples = enc->sample_rate / 200; 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_opus_enc_convert_src (pad, src_fmt, src_val, &dest_fmt, - &dest_val))) - goto error; - gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val); + case 10: + frame_samples = enc->sample_rate / 100; break; - } - case GST_QUERY_LATENCY: - { - gboolean live; - GstClockTime min_latency, max_latency; - gint64 latency; - - if ((res = gst_pad_peer_query (enc->sinkpad, query))) { - gst_query_parse_latency (query, &live, &min_latency, &max_latency); - - latency = gst_opus_enc_get_latency (enc); - - /* add our latency */ - min_latency += latency; - if (max_latency != -1) - max_latency += latency; - - gst_query_set_latency (query, live, min_latency, max_latency); - } + case 20: + frame_samples = enc->sample_rate / 50; + break; + case 40: + frame_samples = enc->sample_rate / 25; + break; + case 60: + frame_samples = 3 * enc->sample_rate / 50; break; - } default: - res = gst_pad_peer_query (pad, query); + GST_WARNING_OBJECT (enc, "Unsupported frame size: %d", enc->frame_size); + frame_samples = 0; break; } - -error: - - gst_object_unref (enc); - - return res; + return frame_samples; } static gboolean -gst_opus_enc_sink_query (GstPad * pad, GstQuery * query) +gst_opus_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info) { - gboolean res = TRUE; + GstOpusEnc *enc; - switch (GST_QUERY_TYPE (query)) { - 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_opus_enc_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; + enc = GST_OPUS_ENC (benc); + + enc->n_channels = GST_AUDIO_INFO_CHANNELS (info); + enc->sample_rate = GST_AUDIO_INFO_RATE (info); + GST_DEBUG_OBJECT (benc, "Setup with %d channels, %d Hz", enc->n_channels, + enc->sample_rate); + + /* handle reconfigure */ + if (enc->state) { + opus_encoder_destroy (enc->state); + enc->state = NULL; } + if (!gst_opus_enc_setup (enc)) + return FALSE; + + enc->frame_samples = gst_opus_enc_get_frame_samples (enc); -error: - return res; + /* feedback to base class */ + gst_audio_encoder_set_latency (benc, + gst_opus_enc_get_latency (enc), gst_opus_enc_get_latency (enc)); + gst_audio_encoder_set_frame_samples_min (benc, + enc->frame_samples * enc->n_channels * 2); + gst_audio_encoder_set_frame_samples_max (benc, + enc->frame_samples * enc->n_channels * 2); + gst_audio_encoder_set_frame_max (benc, 0); + + return TRUE; } -static void -gst_opus_enc_init (GstOpusEnc * enc, GstOpusEncClass * klass) +static GstBuffer * +gst_opus_enc_create_id_buffer (GstOpusEnc * enc) { - enc->sinkpad = gst_pad_new_from_static_template (&sink_factory, "sink"); - gst_element_add_pad (GST_ELEMENT (enc), enc->sinkpad); - gst_pad_set_event_function (enc->sinkpad, - GST_DEBUG_FUNCPTR (gst_opus_enc_sinkevent)); - gst_pad_set_chain_function (enc->sinkpad, - GST_DEBUG_FUNCPTR (gst_opus_enc_chain)); - gst_pad_set_setcaps_function (enc->sinkpad, - GST_DEBUG_FUNCPTR (gst_opus_enc_sink_setcaps)); - gst_pad_set_getcaps_function (enc->sinkpad, - GST_DEBUG_FUNCPTR (gst_opus_enc_sink_getcaps)); - gst_pad_set_query_function (enc->sinkpad, - GST_DEBUG_FUNCPTR (gst_opus_enc_sink_query)); - - enc->srcpad = gst_pad_new_from_static_template (&src_factory, "src"); - gst_pad_set_query_function (enc->srcpad, - GST_DEBUG_FUNCPTR (gst_opus_enc_src_query)); - gst_pad_set_query_type_function (enc->srcpad, - GST_DEBUG_FUNCPTR (gst_opus_enc_get_query_types)); - gst_element_add_pad (GST_ELEMENT (enc), enc->srcpad); + GstBuffer *buffer; + GstByteWriter bw; - enc->n_channels = -1; - enc->sample_rate = -1; - enc->frame_samples = 0; + gst_byte_writer_init (&bw); - enc->bitrate = DEFAULT_BITRATE; - enc->bandwidth = DEFAULT_BANDWIDTH; - enc->frame_size = DEFAULT_FRAMESIZE; - enc->cbr = DEFAULT_CBR; - enc->constrained_vbr = DEFAULT_CONSTRAINED_VBR; - enc->complexity = DEFAULT_COMPLEXITY; - enc->inband_fec = DEFAULT_INBAND_FEC; - enc->dtx = DEFAULT_DTX; - enc->packet_loss_percentage = DEFAULT_PACKET_LOSS_PERCENT; + /* See http://wiki.xiph.org/OggOpus */ + gst_byte_writer_put_string_utf8 (&bw, "OpusHead"); + gst_byte_writer_put_uint8 (&bw, 0); /* version number */ + gst_byte_writer_put_uint8 (&bw, enc->n_channels); + gst_byte_writer_put_uint16_le (&bw, 0); /* pre-skip *//* TODO: endianness ? */ + gst_byte_writer_put_uint32_le (&bw, enc->sample_rate); + gst_byte_writer_put_uint16_le (&bw, 0); /* output gain *//* TODO: endianness ? */ + gst_byte_writer_put_uint8 (&bw, 0); /* channel mapping *//* TODO: what is this ? */ - enc->setup = FALSE; - enc->header_sent = FALSE; + buffer = gst_byte_writer_reset_and_get_buffer (&bw); - enc->adapter = gst_adapter_new (); + GST_BUFFER_OFFSET (buffer) = 0; + GST_BUFFER_OFFSET_END (buffer) = 0; + + return buffer; } -#if 0 static GstBuffer * gst_opus_enc_create_metadata_buffer (GstOpusEnc * enc) { @@ -649,10 +431,11 @@ gst_opus_enc_create_metadata_buffer (GstOpusEnc * enc) empty_tags = gst_tag_list_new (); tags = empty_tags; } - comments = gst_tag_list_to_vorbiscomment_buffer (tags, NULL, - 0, "Encoded with GStreamer Opusenc"); + comments = + gst_tag_list_to_vorbiscomment_buffer (tags, (const guint8 *) "OpusTags", + 8, "Encoded with GStreamer Opusenc"); - GST_BUFFER_OFFSET (comments) = enc->bytes_out; + GST_BUFFER_OFFSET (comments) = 0; GST_BUFFER_OFFSET_END (comments) = 0; if (empty_tags) @@ -660,13 +443,14 @@ gst_opus_enc_create_metadata_buffer (GstOpusEnc * enc) return comments; } -#endif static gboolean gst_opus_enc_setup (GstOpusEnc * enc) { int error = OPUS_OK; + GST_DEBUG_OBJECT (enc, "setup"); + enc->setup = FALSE; enc->state = opus_encoder_create (enc->sample_rate, enc->n_channels, @@ -692,88 +476,20 @@ gst_opus_enc_setup (GstOpusEnc * enc) return TRUE; -#if 0 -mode_initialization_failed: - GST_ERROR_OBJECT (enc, "Mode initialization failed: %d", error); - return FALSE; -#endif - encoder_creation_failed: GST_ERROR_OBJECT (enc, "Encoder creation failed"); return FALSE; } - -/* push out the buffer and do internal bookkeeping */ -static GstFlowReturn -gst_opus_enc_push_buffer (GstOpusEnc * enc, GstBuffer * buffer) -{ - guint size; - - size = GST_BUFFER_SIZE (buffer); - - enc->bytes_out += size; - - GST_DEBUG_OBJECT (enc, "pushing output buffer of size %u", size); - - return gst_pad_push (enc->srcpad, buffer); -} - -#if 0 -static GstCaps * -gst_opus_enc_set_header_on_caps (GstCaps * caps, GstBuffer * buf1, - GstBuffer * buf2) -{ - GstStructure *structure = NULL; - GstBuffer *buf; - GValue array = { 0 }; - GValue value = { 0 }; - - caps = gst_caps_make_writable (caps); - structure = gst_caps_get_structure (caps, 0); - - g_assert (gst_buffer_is_metadata_writable (buf1)); - g_assert (gst_buffer_is_metadata_writable (buf2)); - - /* mark buffers */ - GST_BUFFER_FLAG_SET (buf1, GST_BUFFER_FLAG_IN_CAPS); - GST_BUFFER_FLAG_SET (buf2, GST_BUFFER_FLAG_IN_CAPS); - - /* put buffers in a fixed list */ - g_value_init (&array, GST_TYPE_ARRAY); - g_value_init (&value, GST_TYPE_BUFFER); - buf = gst_buffer_copy (buf1); - gst_value_set_buffer (&value, buf); - gst_buffer_unref (buf); - gst_value_array_append_value (&array, &value); - g_value_unset (&value); - g_value_init (&value, GST_TYPE_BUFFER); - buf = gst_buffer_copy (buf2); - gst_value_set_buffer (&value, buf); - gst_buffer_unref (buf); - gst_value_array_append_value (&array, &value); - gst_structure_set_value (structure, "streamheader", &array); - g_value_unset (&value); - g_value_unset (&array); - - return caps; -} -#endif - - static gboolean -gst_opus_enc_sinkevent (GstPad * pad, GstEvent * event) +gst_opus_enc_sink_event (GstAudioEncoder * benc, GstEvent * event) { - gboolean res = TRUE; GstOpusEnc *enc; - enc = GST_OPUS_ENC (gst_pad_get_parent (pad)); + enc = GST_OPUS_ENC (benc); + GST_DEBUG_OBJECT (enc, "sink event: %s", GST_EVENT_TYPE_NAME (event)); switch (GST_EVENT_TYPE (event)) { - case GST_EVENT_EOS: - gst_opus_enc_encode (enc, TRUE); - res = gst_pad_event_default (pad, event); - break; case GST_EVENT_TAG: { GstTagList *list; @@ -782,62 +498,94 @@ gst_opus_enc_sinkevent (GstPad * pad, GstEvent * event) gst_event_parse_tag (event, &list); gst_tag_setter_merge_tags (setter, list, mode); - res = gst_pad_event_default (pad, event); break; } default: - res = gst_pad_event_default (pad, event); break; } - gst_object_unref (enc); - - return res; + return FALSE; } static GstFlowReturn -gst_opus_enc_encode (GstOpusEnc * enc, gboolean flush) +gst_opus_enc_pre_push (GstAudioEncoder * benc, GstBuffer ** buffer) { - GstFlowReturn ret = GST_FLOW_OK; - gint bytes = enc->frame_samples * 2 * enc->n_channels; - gint bytes_per_packet; + GstOpusEnc *enc; - bytes_per_packet = - (enc->bitrate * enc->frame_samples / enc->sample_rate + 4) / 8; + enc = GST_OPUS_ENC (benc); + + /* FIXME 0.11 ? get rid of this special ogg stuff and have it + * put and use 'codec data' in caps like anything else, + * with all the usual out-of-band advantage etc */ + if (G_UNLIKELY (enc->headers)) { + GSList *header = enc->headers; + + /* try to push all of these, if we lose one, might as well lose all */ + while (header) { + if (ret == GST_FLOW_OK) + ret = gst_pad_push (GST_AUDIO_ENCODER_SRC_PAD (enc), header->data); + else + gst_pad_push (GST_AUDIO_ENCODER_SRC_PAD (enc), header->data); + header = g_slist_next (header); + } - if (flush && gst_adapter_available (enc->adapter) % bytes != 0) { - guint diff = bytes - gst_adapter_available (enc->adapter) % bytes; - GstBuffer *buf = gst_buffer_new_and_alloc (diff); + g_slist_free (enc->headers); + enc->headers = NULL; + } - memset (GST_BUFFER_DATA (buf), 0, diff); - gst_adapter_push (enc->adapter, buf); + return ret; +} + +static GstFlowReturn +gst_opus_enc_encode (GstOpusEnc * enc, GstBuffer * buf) +{ + guint8 *bdata, *data, *mdata = NULL; + gsize bsize, size; + gsize bytes = enc->frame_samples * enc->n_channels * 2; + gsize bytes_per_packet = + (enc->bitrate * enc->frame_samples / enc->sample_rate + 4) / 8; + gint ret = GST_FLOW_OK; + + if (G_LIKELY (buf)) { + bdata = GST_BUFFER_DATA (buf); + bsize = GST_BUFFER_SIZE (buf); + if (G_UNLIKELY (bsize % bytes)) { + GST_DEBUG_OBJECT (enc, "draining; adding silence samples"); + + size = ((bsize / bytes) + 1) * bytes; + mdata = g_malloc0 (size); + memcpy (mdata, bdata, bsize); + bdata = NULL; + data = mdata; + } else { + data = bdata; + size = bsize; + } + } else { + GST_DEBUG_OBJECT (enc, "nothing to drain"); + goto done; } - while (gst_adapter_available (enc->adapter) >= bytes) { - gint16 *data; + while (size) { gint outsize; GstBuffer *outbuf; - ret = gst_pad_alloc_buffer_and_set_caps (enc->srcpad, - GST_BUFFER_OFFSET_NONE, bytes_per_packet, GST_PAD_CAPS (enc->srcpad), - &outbuf); + ret = gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_ENCODER_SRC_PAD (enc), + GST_BUFFER_OFFSET_NONE, bytes_per_packet, + GST_PAD_CAPS (GST_AUDIO_ENCODER_SRC_PAD (enc)), &outbuf); if (GST_FLOW_OK != ret) goto done; - data = (gint16 *) gst_adapter_take (enc->adapter, bytes); - enc->samples_in += enc->frame_samples; - - GST_DEBUG_OBJECT (enc, "encoding %d samples (%d bytes)", - enc->frame_samples, bytes); + GST_DEBUG_OBJECT (enc, "encoding %d samples (%d bytes) to %d bytes", + enc->frame_samples, bytes, bytes_per_packet); - outsize = opus_encode (enc->state, data, enc->frame_samples, + outsize = + opus_encode (enc->state, (const gint16 *) data, enc->frame_samples, GST_BUFFER_DATA (outbuf), bytes_per_packet); - g_free (data); - if (outsize < 0) { GST_ERROR_OBJECT (enc, "Encoding failed: %d", outsize); ret = GST_FLOW_ERROR; @@ -850,149 +598,132 @@ gst_opus_enc_encode (GstOpusEnc * enc, gboolean flush) goto done; } - GST_BUFFER_TIMESTAMP (outbuf) = enc->start_ts + - gst_util_uint64_scale_int (enc->frameno_out * enc->frame_samples, - GST_SECOND, enc->sample_rate); - GST_BUFFER_DURATION (outbuf) = - gst_util_uint64_scale_int (enc->frame_samples, GST_SECOND, - enc->sample_rate); - GST_BUFFER_OFFSET (outbuf) = - gst_util_uint64_scale_int (GST_BUFFER_OFFSET_END (outbuf), GST_SECOND, - enc->sample_rate); - - enc->frameno++; - enc->frameno_out++; - - ret = gst_opus_enc_push_buffer (enc, outbuf); + ret = + gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (enc), outbuf, + enc->frame_samples); if ((GST_FLOW_OK != ret) && (GST_FLOW_NOT_LINKED != ret)) goto done; + + data += bytes; + size -= bytes; } done: + if (mdata) + g_free (mdata); + return ret; } -static GstFlowReturn -gst_opus_enc_chain (GstPad * pad, GstBuffer * buf) +/* + * (really really) FIXME: move into core (dixit tpm) + */ +/** + * _gst_caps_set_buffer_array: + * @caps: a #GstCaps + * @field: field in caps to set + * @buf: header buffers + * + * Adds given buffers to an array of buffers set as the given @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 + */ +static GstCaps * +_gst_caps_set_buffer_array (GstCaps * caps, const gchar * field, + GstBuffer * buf, ...) { - GstOpusEnc *enc; - GstFlowReturn ret = GST_FLOW_OK; - - enc = GST_OPUS_ENC (GST_PAD_PARENT (pad)); + GstStructure *structure = NULL; + va_list va; + GValue array = { 0 }; + GValue value = { 0 }; - if (!enc->setup) - goto not_setup; + g_return_val_if_fail (caps != NULL, NULL); + g_return_val_if_fail (gst_caps_is_fixed (caps), NULL); + g_return_val_if_fail (field != NULL, NULL); - if (!enc->header_sent) { - GstCaps *caps; + caps = gst_caps_make_writable (caps); + structure = gst_caps_get_structure (caps, 0); - caps = gst_pad_get_caps (enc->srcpad); - gst_caps_set_simple (caps, - "rate", G_TYPE_INT, enc->sample_rate, - "channels", G_TYPE_INT, enc->n_channels, - "frame-size", G_TYPE_INT, enc->frame_size, NULL); + g_value_init (&array, GST_TYPE_ARRAY); - /* negotiate with these caps */ - GST_DEBUG_OBJECT (enc, "here are the caps: %" GST_PTR_FORMAT, caps); - GST_LOG_OBJECT (enc, "rate=%d channels=%d frame-size=%d", - enc->sample_rate, enc->n_channels, enc->frame_size); - gst_pad_set_caps (enc->srcpad, caps); + va_start (va, buf); + /* put buffers in a fixed list */ + while (buf) { + g_assert (gst_buffer_is_writable (buf)); - enc->header_sent = TRUE; - } + /* mark buffer */ + GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_IN_CAPS); - GST_DEBUG_OBJECT (enc, "received buffer of %u bytes", GST_BUFFER_SIZE (buf)); + 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); - /* Save the timestamp of the first buffer. This will be later - * used as offset for all following buffers */ - if (enc->start_ts == GST_CLOCK_TIME_NONE) { - if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)) { - enc->start_ts = GST_BUFFER_TIMESTAMP (buf); - } else { - enc->start_ts = 0; - } + buf = va_arg (va, GstBuffer *); } + gst_structure_set_value (structure, field, &array); + g_value_unset (&array); - /* Check if we have a continous stream, if not drop some samples or the buffer or - * insert some silence samples */ - if (enc->next_ts != GST_CLOCK_TIME_NONE && - GST_BUFFER_TIMESTAMP (buf) < enc->next_ts) { - guint64 diff = enc->next_ts - GST_BUFFER_TIMESTAMP (buf); - guint64 diff_bytes; - - GST_WARNING_OBJECT (enc, "Buffer is older than previous " - "timestamp + duration (%" GST_TIME_FORMAT "< %" GST_TIME_FORMAT - "), cannot handle. Clipping buffer.", - GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)), - GST_TIME_ARGS (enc->next_ts)); - - diff_bytes = - GST_CLOCK_TIME_TO_FRAMES (diff, enc->sample_rate) * enc->n_channels * 2; - if (diff_bytes >= GST_BUFFER_SIZE (buf)) { - gst_buffer_unref (buf); - return GST_FLOW_OK; - } - buf = gst_buffer_make_metadata_writable (buf); - GST_BUFFER_DATA (buf) += diff_bytes; - GST_BUFFER_SIZE (buf) -= diff_bytes; + return caps; +} - GST_BUFFER_TIMESTAMP (buf) += diff; - if (GST_BUFFER_DURATION_IS_VALID (buf)) - GST_BUFFER_DURATION (buf) -= diff; - } +static GstFlowReturn +gst_opus_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf) +{ + GstOpusEnc *enc; + GstFlowReturn ret = GST_FLOW_OK; - if (enc->next_ts != GST_CLOCK_TIME_NONE - && GST_BUFFER_TIMESTAMP_IS_VALID (buf)) { - guint64 max_diff = - gst_util_uint64_scale (enc->frame_size, GST_SECOND, enc->sample_rate); + enc = GST_OPUS_ENC (benc); + GST_DEBUG_OBJECT (enc, "handle_frame"); - if (GST_BUFFER_TIMESTAMP (buf) != enc->next_ts && - GST_BUFFER_TIMESTAMP (buf) - enc->next_ts > max_diff) { - GST_WARNING_OBJECT (enc, - "Discontinuity detected: %" G_GUINT64_FORMAT " > %" G_GUINT64_FORMAT, - GST_BUFFER_TIMESTAMP (buf) - enc->next_ts, max_diff); + if (!enc->header_sent) { + /* Opus streams in Ogg begin with two headers; the initial header (with + most of the codec setup parameters) which is mandated by the Ogg + bitstream spec. The second header holds any comment fields. */ + GstBuffer *buf1, *buf2; + GstCaps *caps; - gst_opus_enc_encode (enc, TRUE); + /* create header buffers */ + buf1 = gst_opus_enc_create_id_buffer (enc); + buf2 = gst_opus_enc_create_metadata_buffer (enc); - enc->frameno_out = 0; - enc->start_ts = GST_BUFFER_TIMESTAMP (buf); - } - } + /* mark and put on caps */ + caps = + gst_caps_new_simple ("audio/x-opus", "rate", G_TYPE_INT, + enc->sample_rate, "channels", G_TYPE_INT, enc->n_channels, "frame-size", + G_TYPE_INT, enc->frame_size, NULL); + caps = _gst_caps_set_buffer_array (caps, "streamheader", buf1, buf2, NULL); - if (GST_BUFFER_TIMESTAMP_IS_VALID (buf) - && GST_BUFFER_DURATION_IS_VALID (buf)) - enc->next_ts = GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf); - else - enc->next_ts = GST_CLOCK_TIME_NONE; + /* negotiate with these caps */ + GST_DEBUG_OBJECT (enc, "here are the caps: %" GST_PTR_FORMAT, caps); - /* push buffer to adapter */ - gst_adapter_push (enc->adapter, buf); - buf = NULL; + gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (enc), caps); - ret = gst_opus_enc_encode (enc, FALSE); + /* push out buffers */ + /* store buffers for later pre_push sending */ + g_slist_foreach (enc->headers, (GFunc) gst_buffer_unref, NULL); + enc->headers = NULL; + GST_DEBUG_OBJECT (enc, "storing header buffers"); + enc->headers = g_slist_prepend (enc->headers, buf2); + enc->headers = g_slist_prepend (enc->headers, buf1); + enc->header_sent = TRUE; + } -done: + GST_DEBUG_OBJECT (enc, "received buffer %p of %u bytes", buf, + buf ? GST_BUFFER_SIZE (buf) : 0); - if (buf) - gst_buffer_unref (buf); + ret = gst_opus_enc_encode (enc, buf); return ret; - - /* ERRORS */ -not_setup: - { - GST_ELEMENT_ERROR (enc, CORE, NEGOTIATION, (NULL), - ("encoder not initialized (input is not audio?)")); - ret = GST_FLOW_NOT_NEGOTIATED; - goto done; - } - } - static void gst_opus_enc_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) @@ -1082,49 +813,3 @@ gst_opus_enc_set_property (GObject * object, guint prop_id, break; } } - -static GstStateChangeReturn -gst_opus_enc_change_state (GstElement * element, GstStateChange transition) -{ - GstOpusEnc *enc = GST_OPUS_ENC (element); - GstStateChangeReturn res; - - switch (transition) { - case GST_STATE_CHANGE_NULL_TO_READY: - break; - case GST_STATE_CHANGE_READY_TO_PAUSED: - enc->frameno = 0; - enc->samples_in = 0; - enc->frameno_out = 0; - enc->start_ts = GST_CLOCK_TIME_NONE; - enc->next_ts = GST_CLOCK_TIME_NONE; - break; - case GST_STATE_CHANGE_PAUSED_TO_PLAYING: - /* fall through */ - default: - break; - } - - res = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); - if (res == GST_STATE_CHANGE_FAILURE) - return res; - - switch (transition) { - case GST_STATE_CHANGE_PLAYING_TO_PAUSED: - break; - case GST_STATE_CHANGE_PAUSED_TO_READY: - enc->setup = FALSE; - enc->header_sent = FALSE; - if (enc->state) { - opus_encoder_destroy (enc->state); - enc->state = NULL; - } - break; - case GST_STATE_CHANGE_READY_TO_NULL: - gst_tag_setter_reset_tags (GST_TAG_SETTER (enc)); - default: - break; - } - - return res; -} |