From ffc4408d539541ff50080e1ae2665bf7851f4a77 Mon Sep 17 00:00:00 2001 From: Vincent Penquerc'h Date: Wed, 16 Nov 2011 17:05:17 +0000 Subject: [PATCH] opusdec: rewrite logic Parameters such as frame size, etc, are variable. Pretty much everything can change within a stream, so be prepared about it, and do not cache parameters in the decoder. --- ext/opus/gstopusdec.c | 123 +++++++++++++++++++------------------------------- ext/opus/gstopusdec.h | 4 +- 2 files changed, 47 insertions(+), 80 deletions(-) diff --git a/ext/opus/gstopusdec.c b/ext/opus/gstopusdec.c index 1abe320..f14e4e0 100644 --- a/ext/opus/gstopusdec.c +++ b/ext/opus/gstopusdec.c @@ -48,8 +48,6 @@ GST_DEBUG_CATEGORY_STATIC (opusdec_debug); #define GST_CAT_DEFAULT opusdec_debug -#define DEC_MAX_FRAME_SIZE 2000 - static GstStaticPadTemplate opus_dec_src_factory = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, @@ -117,14 +115,13 @@ static void gst_opus_dec_reset (GstOpusDec * dec) { dec->packetno = 0; - dec->frame_size = 0; - dec->frame_samples = 960; - dec->frame_duration = 0; if (dec->state) { opus_decoder_destroy (dec->state); dec->state = NULL; } + dec->next_ts = 0; + gst_buffer_replace (&dec->streamheader, NULL); gst_buffer_replace (&dec->vorbiscomment, NULL); } @@ -183,6 +180,8 @@ opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buf, GstBuffer *outbuf; gint16 *out_data; int n, err; + int samples; + unsigned int packet_size; if (dec->state == NULL) { GstCaps *caps; @@ -199,8 +198,8 @@ opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buf, "endianness", G_TYPE_INT, G_BYTE_ORDER, "width", G_TYPE_INT, 16, "depth", G_TYPE_INT, 16, NULL); - GST_DEBUG_OBJECT (dec, "rate=%d channels=%d frame-size=%d", - dec->sample_rate, dec->n_channels, dec->frame_size); + GST_DEBUG_OBJECT (dec, "rate=%d channels=%d", + dec->sample_rate, dec->n_channels); if (!gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), caps)) GST_ERROR ("nego failure"); @@ -222,14 +221,15 @@ opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buf, size = 0; } - GST_DEBUG ("frames %d", opus_packet_get_nb_frames (data, size)); + samples = + opus_packet_get_samples_per_frame (data, + dec->sample_rate) * opus_packet_get_nb_frames (data, size); + packet_size = samples * dec->n_channels * 2; GST_DEBUG ("bandwidth %d", opus_packet_get_bandwidth (data)); - GST_DEBUG ("samples_per_frame %d", opus_packet_get_samples_per_frame (data, - 48000)); - GST_DEBUG ("channels %d", opus_packet_get_nb_channels (data)); + GST_DEBUG ("samples %d", samples); res = gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), - GST_BUFFER_OFFSET_NONE, dec->frame_samples * dec->n_channels * 2, + GST_BUFFER_OFFSET_NONE, packet_size, GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (dec)), &outbuf); if (res != GST_FLOW_OK) { @@ -239,27 +239,28 @@ opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buf, out_data = (gint16 *) GST_BUFFER_DATA (outbuf); - GST_LOG_OBJECT (dec, "decoding %d sample frame", dec->frame_samples); + GST_LOG_OBJECT (dec, "decoding %d samples, in size %u", samples, size); - n = opus_decode (dec->state, data, size, out_data, dec->frame_samples, 0); + n = opus_decode (dec->state, data, size, out_data, samples, 0); if (n < 0) { GST_ELEMENT_ERROR (dec, STREAM, DECODE, ("Decoding error: %d", n), (NULL)); return GST_FLOW_ERROR; } + GST_DEBUG_OBJECT (dec, "decoded %d samples", n); - if (!GST_CLOCK_TIME_IS_VALID (timestamp)) { - GST_WARNING_OBJECT (dec, "No timestamp in -> no timestamp out"); + if (GST_CLOCK_TIME_IS_VALID (timestamp)) { + GST_BUFFER_TIMESTAMP (outbuf) = timestamp; + } else { + GST_BUFFER_TIMESTAMP (outbuf) = dec->next_ts; } - GST_DEBUG_OBJECT (dec, "timestamp=%" GST_TIME_FORMAT, - GST_TIME_ARGS (timestamp)); - - GST_BUFFER_TIMESTAMP (outbuf) = GST_BUFFER_TIMESTAMP (buf); - GST_BUFFER_DURATION (outbuf) = GST_BUFFER_DURATION (buf); + GST_BUFFER_DURATION (outbuf) = + gst_util_uint64_scale (n, GST_SECOND, dec->sample_rate); + dec->next_ts = GST_BUFFER_TIMESTAMP (outbuf) + GST_BUFFER_DURATION (outbuf); GST_LOG_OBJECT (dec, "pushing buffer with ts=%" GST_TIME_FORMAT ", dur=%" GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)), - GST_TIME_ARGS (dec->frame_duration)); + GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf))); res = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), outbuf, 1); @@ -273,37 +274,6 @@ creation_failed: return GST_FLOW_ERROR; } -static gint -gst_opus_dec_get_frame_samples (GstOpusDec * dec) -{ - gint frame_samples = 0; - switch (dec->frame_size) { - case 2: - frame_samples = dec->sample_rate / 400; - break; - case 5: - frame_samples = dec->sample_rate / 200; - break; - case 10: - frame_samples = dec->sample_rate / 100; - break; - case 20: - frame_samples = dec->sample_rate / 50; - break; - case 40: - frame_samples = dec->sample_rate / 25; - break; - case 60: - frame_samples = 3 * dec->sample_rate / 50; - break; - default: - GST_WARNING_OBJECT (dec, "Unsupported frame size: %d", dec->frame_size); - frame_samples = 0; - break; - } - return frame_samples; -} - static gboolean gst_opus_dec_set_format (GstAudioDecoder * bdec, GstCaps * caps) { @@ -340,24 +310,6 @@ gst_opus_dec_set_format (GstAudioDecoder * bdec, GstCaps * caps) gst_buffer_replace (&dec->vorbiscomment, buf); } } - if (!gst_structure_get_int (s, "frame-size", &dec->frame_size)) { - GST_WARNING_OBJECT (dec, "Frame size not included in caps"); - } - if (!gst_structure_get_int (s, "channels", &dec->n_channels)) { - GST_WARNING_OBJECT (dec, "Number of channels not included in caps"); - } - if (!gst_structure_get_int (s, "rate", &dec->sample_rate)) { - GST_WARNING_OBJECT (dec, "Sample rate not included in caps"); - } - - dec->frame_samples = gst_opus_dec_get_frame_samples (dec); - dec->frame_duration = gst_util_uint64_scale_int (dec->frame_samples, - GST_SECOND, dec->sample_rate); - GST_INFO_OBJECT (dec, - "Got frame size %d, %d channels, %d Hz, giving %d samples per frame, frame duration %" - GST_TIME_FORMAT, dec->frame_size, dec->n_channels, dec->sample_rate, - dec->frame_samples, GST_TIME_ARGS (dec->frame_duration)); - caps = gst_caps_new_simple ("audio/x-raw-int", "rate", G_TYPE_INT, dec->sample_rate, "channels", G_TYPE_INT, dec->n_channels, @@ -385,6 +337,13 @@ memcmp_buffers (GstBuffer * buf1, GstBuffer * buf2) return !memcmp (GST_BUFFER_DATA (buf1), GST_BUFFER_DATA (buf2), size1); } +static gboolean +gst_opus_dec_is_header (GstBuffer * buf, const char *magic, guint magic_size) +{ + return (GST_BUFFER_SIZE (buf) >= magic_size + && !memcmp (magic, GST_BUFFER_DATA (buf), magic_size)); +} + static GstFlowReturn gst_opus_dec_handle_frame (GstAudioDecoder * adec, GstBuffer * buf) { @@ -421,14 +380,24 @@ gst_opus_dec_handle_frame (GstAudioDecoder * adec, GstBuffer * buf) * first two packets are the headers. */ switch (dec->packetno) { case 0: - GST_DEBUG_OBJECT (dec, "counted streamheader"); - res = gst_opus_dec_parse_header (dec, buf); - gst_audio_decoder_finish_frame (adec, NULL, 1); + if (gst_opus_dec_is_header (buf, "OpusHead", 8)) { + GST_DEBUG_OBJECT (dec, "found streamheader"); + res = gst_opus_dec_parse_header (dec, buf); + gst_audio_decoder_finish_frame (adec, NULL, 1); + } else { + res = opus_dec_chain_parse_data (dec, buf, GST_BUFFER_TIMESTAMP (buf), + GST_BUFFER_DURATION (buf)); + } break; case 1: - GST_DEBUG_OBJECT (dec, "counted vorbiscomments"); - res = gst_opus_dec_parse_comments (dec, buf); - gst_audio_decoder_finish_frame (adec, NULL, 1); + if (gst_opus_dec_is_header (buf, "OpusTags", 8)) { + GST_DEBUG_OBJECT (dec, "counted vorbiscomments"); + res = gst_opus_dec_parse_comments (dec, buf); + gst_audio_decoder_finish_frame (adec, NULL, 1); + } else { + res = opus_dec_chain_parse_data (dec, buf, GST_BUFFER_TIMESTAMP (buf), + GST_BUFFER_DURATION (buf)); + } break; default: { diff --git a/ext/opus/gstopusdec.h b/ext/opus/gstopusdec.h index 8389580..38dd279 100644 --- a/ext/opus/gstopusdec.h +++ b/ext/opus/gstopusdec.h @@ -45,11 +45,9 @@ struct _GstOpusDec { GstAudioDecoder element; OpusDecoder *state; - int frame_samples; - gint frame_size; - GstClockTime frame_duration; guint64 packetno; + GstClockTime next_ts; GstBuffer *streamheader; GstBuffer *vorbiscomment; -- 2.7.4