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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-25 01:30:38 +00:00
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.
This commit is contained in:
Vincent Penquerc'h
parent
9690361d04
commit
afbc4fdd71
2 changed files with 53 additions and 87 deletions
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@ -48,9 +48,6 @@
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GST_DEBUG_CATEGORY_STATIC (opusdec_debug);
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#define GST_CAT_DEFAULT opusdec_debug
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#define DEC_MAX_FRAME_SIZE 2000
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#define DEC_MAX_OUTPUT_BUFFER_SIZE (5760)
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static GstStaticPadTemplate opus_dec_src_factory =
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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@ -110,14 +107,13 @@ static void
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gst_opus_dec_reset (GstOpusDec * dec)
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{
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dec->packetno = 0;
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dec->frame_size = 0;
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dec->frame_samples = 960;
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dec->frame_duration = 0;
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if (dec->state) {
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opus_decoder_destroy (dec->state);
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dec->state = NULL;
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}
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dec->next_ts = 0;
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gst_buffer_replace (&dec->streamheader, NULL);
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gst_buffer_replace (&dec->vorbiscomment, NULL);
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}
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@ -176,7 +172,7 @@ opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buf,
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GstBuffer *outbuf;
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gint16 *out_data;
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int n, err;
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int samples_per_frame;
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int samples;
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unsigned int packet_size;
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if (dec->state == NULL) {
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@ -192,8 +188,8 @@ opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buf,
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"rate", G_TYPE_INT, dec->sample_rate,
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"channels", G_TYPE_INT, dec->n_channels, NULL);
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GST_DEBUG_OBJECT (dec, "rate=%d channels=%d frame-size=%d",
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dec->sample_rate, dec->n_channels, dec->frame_size);
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GST_DEBUG_OBJECT (dec, "rate=%d channels=%d",
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dec->sample_rate, dec->n_channels);
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if (!gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), caps))
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GST_ERROR ("nego failure");
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@ -214,12 +210,13 @@ opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buf,
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size = 0;
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}
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samples_per_frame =
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opus_packet_get_samples_per_frame (data, dec->sample_rate);
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samples =
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opus_packet_get_samples_per_frame (data,
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dec->sample_rate) * opus_packet_get_nb_frames (data, size);
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GST_DEBUG ("bandwidth %d", opus_packet_get_bandwidth (data));
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GST_DEBUG ("samples_per_frame %d", samples_per_frame);
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GST_DEBUG ("samples %d", samples);
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packet_size = samples_per_frame * dec->n_channels * 2;
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packet_size = samples * dec->n_channels * 2;
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outbuf = gst_buffer_new_and_alloc (packet_size);
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if (!outbuf) {
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goto buffer_failed;
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@ -227,34 +224,33 @@ opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buf,
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out_data = (gint16 *) gst_buffer_map (outbuf, &out_size, NULL, GST_MAP_WRITE);
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GST_LOG_OBJECT (dec, "decoding frame");
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GST_LOG_OBJECT (dec, "decoding %d samples, in size %u", samples, size);
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n = opus_decode (dec->state, data, size, out_data, dec->frame_samples, 0);
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n = opus_decode (dec->state, data, size, out_data, samples, 0);
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gst_buffer_unmap (buf, data, size);
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gst_buffer_unmap (outbuf, out_data, out_size);
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if (n < 0) {
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gst_buffer_unmap (outbuf, out_data, out_size);
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GST_ELEMENT_ERROR (dec, STREAM, DECODE, ("Decoding error: %d", n), (NULL));
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return GST_FLOW_ERROR;
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}
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GST_DEBUG_OBJECT (dec, "decoded %d samples", n);
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if (!GST_CLOCK_TIME_IS_VALID (timestamp)) {
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GST_WARNING_OBJECT (dec, "No timestamp in -> no timestamp out");
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if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
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GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
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} else {
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GST_BUFFER_TIMESTAMP (outbuf) = dec->next_ts;
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}
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GST_DEBUG_OBJECT (dec, "timestamp=%" GST_TIME_FORMAT,
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GST_TIME_ARGS (timestamp));
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GST_BUFFER_TIMESTAMP (outbuf) = GST_BUFFER_TIMESTAMP (buf);
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GST_BUFFER_DURATION (outbuf) = GST_BUFFER_DURATION (buf);
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GST_BUFFER_DURATION (outbuf) =
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gst_util_uint64_scale (n, GST_SECOND, dec->sample_rate);
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dec->next_ts = GST_BUFFER_TIMESTAMP (outbuf) + GST_BUFFER_DURATION (outbuf);
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GST_LOG_OBJECT (dec, "pushing buffer with ts=%" GST_TIME_FORMAT ", dur=%"
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GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
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GST_TIME_ARGS (dec->frame_duration));
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GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)));
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res = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), outbuf, 1);
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gst_buffer_unmap (outbuf, out_data, out_size);
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if (res != GST_FLOW_OK)
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GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (res));
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@ -269,37 +265,6 @@ buffer_failed:
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return GST_FLOW_ERROR;
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}
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static gint
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gst_opus_dec_get_frame_samples (GstOpusDec * dec)
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{
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gint frame_samples = 0;
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switch (dec->frame_size) {
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case 2:
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frame_samples = dec->sample_rate / 400;
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break;
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case 5:
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frame_samples = dec->sample_rate / 200;
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break;
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case 10:
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frame_samples = dec->sample_rate / 100;
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break;
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case 20:
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frame_samples = dec->sample_rate / 50;
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break;
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case 40:
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frame_samples = dec->sample_rate / 25;
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break;
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case 60:
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frame_samples = 3 * dec->sample_rate / 50;
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break;
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default:
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GST_WARNING_OBJECT (dec, "Unsupported frame size: %d", dec->frame_size);
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frame_samples = 0;
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break;
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}
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return frame_samples;
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}
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static gboolean
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gst_opus_dec_set_format (GstAudioDecoder * bdec, GstCaps * caps)
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{
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@ -337,25 +302,6 @@ gst_opus_dec_set_format (GstAudioDecoder * bdec, GstCaps * caps)
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}
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}
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if (!gst_structure_get_int (s, "frame-size", &dec->frame_size)) {
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GST_WARNING_OBJECT (dec, "Frame size not included in caps");
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}
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if (!gst_structure_get_int (s, "channels", &dec->n_channels)) {
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GST_WARNING_OBJECT (dec, "Number of channels not included in caps");
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}
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if (!gst_structure_get_int (s, "rate", &dec->sample_rate)) {
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GST_WARNING_OBJECT (dec, "Sample rate not included in caps");
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}
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dec->frame_samples = gst_opus_dec_get_frame_samples (dec);
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dec->frame_duration = gst_util_uint64_scale_int (dec->frame_samples,
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GST_SECOND, dec->sample_rate);
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GST_INFO_OBJECT (dec,
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"Got frame size %d, %d channels, %d Hz, giving %d samples per frame, frame duration %"
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GST_TIME_FORMAT, dec->frame_size, dec->n_channels, dec->sample_rate,
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dec->frame_samples, GST_TIME_ARGS (dec->frame_duration));
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caps = gst_caps_new_simple ("audio/x-raw",
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"format", G_TYPE_STRING, "S16LE",
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"rate", G_TYPE_INT, dec->sample_rate,
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@ -387,6 +333,20 @@ memcmp_buffers (GstBuffer * buf1, GstBuffer * buf2)
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return res;
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}
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static gboolean
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gst_opus_dec_is_header (GstBuffer * buf, const char *magic, guint magic_size)
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{
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guint8 *data;
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gsize size;
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gboolean ret;
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data = gst_buffer_map (buf, &size, NULL, GST_MAP_READ);
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if (!data)
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return FALSE;
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ret = (size >= magic_size && !memcmp (magic, data, magic_size));
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gst_buffer_unmap (buf, data, size);
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return ret;
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}
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static GstFlowReturn
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gst_opus_dec_handle_frame (GstAudioDecoder * adec, GstBuffer * buf)
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@ -424,16 +384,24 @@ gst_opus_dec_handle_frame (GstAudioDecoder * adec, GstBuffer * buf)
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* first two packets are the headers. */
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switch (dec->packetno) {
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case 0:
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GST_DEBUG_OBJECT (dec, "counted streamheader");
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res = GST_FLOW_OK;
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res = gst_opus_dec_parse_header (dec, buf);
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gst_audio_decoder_finish_frame (adec, NULL, 1);
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if (gst_opus_dec_is_header (buf, "OpusHead", 8)) {
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GST_DEBUG_OBJECT (dec, "found streamheader");
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res = gst_opus_dec_parse_header (dec, buf);
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gst_audio_decoder_finish_frame (adec, NULL, 1);
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} else {
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res = opus_dec_chain_parse_data (dec, buf, GST_BUFFER_TIMESTAMP (buf),
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GST_BUFFER_DURATION (buf));
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}
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break;
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case 1:
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GST_DEBUG_OBJECT (dec, "counted vorbiscomments");
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res = GST_FLOW_OK;
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res = gst_opus_dec_parse_comments (dec, buf);
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gst_audio_decoder_finish_frame (adec, NULL, 1);
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if (gst_opus_dec_is_header (buf, "OpusTags", 8)) {
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GST_DEBUG_OBJECT (dec, "counted vorbiscomments");
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res = gst_opus_dec_parse_comments (dec, buf);
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gst_audio_decoder_finish_frame (adec, NULL, 1);
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} else {
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res = opus_dec_chain_parse_data (dec, buf, GST_BUFFER_TIMESTAMP (buf),
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GST_BUFFER_DURATION (buf));
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}
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break;
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default:
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{
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@ -45,11 +45,9 @@ struct _GstOpusDec {
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GstAudioDecoder element;
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OpusDecoder *state;
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int frame_samples;
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gint frame_size;
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GstClockTime frame_duration;
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guint64 packetno;
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GstClockTime next_ts;
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GstBuffer *streamheader;
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GstBuffer *vorbiscomment;
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