From afbc4fdd713a939b4f1d893b4048438617f6cdce Mon Sep 17 00:00:00 2001 From: Vincent Penquerc'h Date: Wed, 16 Nov 2011 01:14:32 +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 | 136 ++++++++++++++++-------------------------- ext/opus/gstopusdec.h | 4 +- 2 files changed, 53 insertions(+), 87 deletions(-) diff --git a/ext/opus/gstopusdec.c b/ext/opus/gstopusdec.c index 0b30c38d7a..d3528c85a9 100644 --- a/ext/opus/gstopusdec.c +++ b/ext/opus/gstopusdec.c @@ -48,9 +48,6 @@ GST_DEBUG_CATEGORY_STATIC (opusdec_debug); #define GST_CAT_DEFAULT opusdec_debug -#define DEC_MAX_FRAME_SIZE 2000 -#define DEC_MAX_OUTPUT_BUFFER_SIZE (5760) - static GstStaticPadTemplate opus_dec_src_factory = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, @@ -110,14 +107,13 @@ 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); } @@ -176,7 +172,7 @@ opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buf, GstBuffer *outbuf; gint16 *out_data; int n, err; - int samples_per_frame; + int samples; unsigned int packet_size; if (dec->state == NULL) { @@ -192,8 +188,8 @@ opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buf, "rate", G_TYPE_INT, dec->sample_rate, "channels", G_TYPE_INT, dec->n_channels, 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"); @@ -214,12 +210,13 @@ opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buf, size = 0; } - samples_per_frame = - opus_packet_get_samples_per_frame (data, dec->sample_rate); + samples = + opus_packet_get_samples_per_frame (data, + dec->sample_rate) * opus_packet_get_nb_frames (data, size); GST_DEBUG ("bandwidth %d", opus_packet_get_bandwidth (data)); - GST_DEBUG ("samples_per_frame %d", samples_per_frame); + GST_DEBUG ("samples %d", samples); - packet_size = samples_per_frame * dec->n_channels * 2; + packet_size = samples * dec->n_channels * 2; outbuf = gst_buffer_new_and_alloc (packet_size); if (!outbuf) { goto buffer_failed; @@ -227,34 +224,33 @@ opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buf, out_data = (gint16 *) gst_buffer_map (outbuf, &out_size, NULL, GST_MAP_WRITE); - GST_LOG_OBJECT (dec, "decoding frame"); + 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); gst_buffer_unmap (buf, data, size); + gst_buffer_unmap (outbuf, out_data, out_size); if (n < 0) { - gst_buffer_unmap (outbuf, out_data, out_size); 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); - gst_buffer_unmap (outbuf, out_data, out_size); - if (res != GST_FLOW_OK) GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (res)); @@ -269,37 +265,6 @@ buffer_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) { @@ -337,25 +302,6 @@ gst_opus_dec_set_format (GstAudioDecoder * bdec, GstCaps * caps) } } - 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", "format", G_TYPE_STRING, "S16LE", "rate", G_TYPE_INT, dec->sample_rate, @@ -387,6 +333,20 @@ memcmp_buffers (GstBuffer * buf1, GstBuffer * buf2) return res; } +static gboolean +gst_opus_dec_is_header (GstBuffer * buf, const char *magic, guint magic_size) +{ + guint8 *data; + gsize size; + gboolean ret; + + data = gst_buffer_map (buf, &size, NULL, GST_MAP_READ); + if (!data) + return FALSE; + ret = (size >= magic_size && !memcmp (magic, data, magic_size)); + gst_buffer_unmap (buf, data, size); + return ret; +} static GstFlowReturn gst_opus_dec_handle_frame (GstAudioDecoder * adec, GstBuffer * buf) @@ -424,16 +384,24 @@ 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_FLOW_OK; - 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_FLOW_OK; - 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 8389580e3b..38dd2799ad 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;