mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2025-02-27 08:26:39 +00:00
Merge branch 'master' into 0.11
This commit is contained in:
Wim Taymans
commit
6bd65058c4
2 changed files with 76 additions and 123 deletions
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@ -55,7 +55,7 @@ GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw-int, "
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"rate = (int) [ 32000, 64000 ], "
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"rate = (int) { 8000, 12000, 16000, 24000, 48000 }, "
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"channels = (int) [ 1, 2 ], "
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"endianness = (int) BYTE_ORDER, "
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"signed = (boolean) true, " "width = (int) 16, " "depth = (int) 16")
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@ -193,6 +193,8 @@ opus_dec_sink_setcaps (GstPad * pad, GstCaps * caps)
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GstStructure *s;
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const GValue *streamheader;
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GST_DEBUG_OBJECT (pad, "Setting sink caps to %" GST_PTR_FORMAT, caps);
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s = gst_caps_get_structure (caps, 0);
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if ((streamheader = gst_structure_get_value (s, "streamheader")) &&
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G_VALUE_HOLDS (streamheader, GST_TYPE_ARRAY) &&
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@ -237,6 +239,47 @@ opus_dec_sink_setcaps (GstPad * pad, 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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switch (dec->frame_size) {
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case 2:
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dec->frame_samples = dec->sample_rate / 400;
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break;
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case 5:
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dec->frame_samples = dec->sample_rate / 200;
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break;
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case 10:
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dec->frame_samples = dec->sample_rate / 100;
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break;
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case 20:
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dec->frame_samples = dec->sample_rate / 50;
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break;
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case 40:
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dec->frame_samples = dec->sample_rate / 25;
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break;
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case 60:
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dec->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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break;
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}
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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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done:
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gst_object_unref (dec);
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return ret;
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@ -566,7 +609,7 @@ static GstFlowReturn
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opus_dec_chain_parse_header (GstOpusDec * dec, GstBuffer * buf)
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{
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GstCaps *caps;
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//gint error = OPUS_OK;
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int err;
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#if 0
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dec->samples_per_frame = opus_packet_get_samples_per_frame (
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@ -578,42 +621,10 @@ opus_dec_chain_parse_header (GstOpusDec * dec, GstBuffer * buf)
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goto invalid_header;
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#endif
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#if 0
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#ifdef HAVE_OPUS_0_7
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dec->mode =
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opus_mode_create (dec->sample_rate, dec->header.frame_size, &error);
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#else
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dec->mode =
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opus_mode_create (dec->sample_rate, dec->header.nb_channels,
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dec->header.frame_size, &error);
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#endif
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if (!dec->mode)
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goto mode_init_failed;
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/* initialize the decoder */
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#ifdef HAVE_OPUS_0_11
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dec->state =
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opus_decoder_create_custom (dec->mode, dec->header.nb_channels, &error);
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#else
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#ifdef HAVE_OPUS_0_7
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dec->state = opus_decoder_create (dec->mode, dec->header.nb_channels, &error);
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#else
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dec->state = opus_decoder_create (dec->mode);
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#endif
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#endif
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#endif
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dec->state = opus_decoder_create (dec->sample_rate, dec->n_channels);
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if (!dec->state)
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dec->state = opus_decoder_create (dec->sample_rate, dec->n_channels, &err);
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if (!dec->state || err != OPUS_OK)
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goto init_failed;
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#if 0
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#ifdef HAVE_OPUS_0_8
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dec->frame_size = dec->header.frame_size;
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#else
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opus_mode_info (dec->mode, OPUS_GET_FRAME_SIZE, &dec->frame_size);
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#endif
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#endif
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dec->frame_duration = gst_util_uint64_scale_int (dec->frame_size,
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GST_SECOND, dec->sample_rate);
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@ -711,7 +722,7 @@ opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buf,
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guint8 *data;
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GstBuffer *outbuf;
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gint16 *out_data;
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int n;
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int n, err;
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if (timestamp != -1) {
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dec->segment.last_stop = timestamp;
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@ -721,7 +732,9 @@ opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buf,
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if (dec->state == NULL) {
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GstCaps *caps;
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dec->state = opus_decoder_create (dec->sample_rate, dec->n_channels);
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dec->state = opus_decoder_create (dec->sample_rate, dec->n_channels, &err);
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if (!dec->state || err != OPUS_OK)
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goto creation_failed;
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/* set caps */
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caps = gst_caps_new_simple ("audio/x-raw-int",
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@ -772,7 +785,7 @@ opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buf,
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GST_LOG_OBJECT (dec, "decoding frame");
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n = opus_decode (dec->state, data, size, out_data, dec->frame_samples, TRUE);
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n = opus_decode (dec->state, data, size, out_data, dec->frame_samples, 0);
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if (n < 0) {
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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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@ -805,6 +818,10 @@ opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buf,
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GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (res));
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return res;
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creation_failed:
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GST_ERROR_OBJECT (dec, "Failed to create Opus decoder: %d", err);
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return GST_FLOW_ERROR;
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}
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static GstFlowReturn
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@ -814,6 +831,10 @@ opus_dec_chain (GstPad * pad, GstBuffer * buf)
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GstOpusDec *dec;
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dec = GST_OPUS_DEC (gst_pad_get_parent (pad));
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GST_LOG_OBJECT (pad,
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"Got buffer ts %" GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT,
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GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
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GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
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if (GST_BUFFER_IS_DISCONT (buf)) {
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dec->discont = TRUE;
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@ -63,7 +63,7 @@ gst_opus_enc_bandwidth_get_type (void)
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{OPUS_BANDWIDTH_WIDEBAND, "Wide band", "wideband"},
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{OPUS_BANDWIDTH_SUPERWIDEBAND, "Super wide band", "superwideband"},
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{OPUS_BANDWIDTH_FULLBAND, "Full band", "fullband"},
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{OPUS_BANDWIDTH_AUTO, "Auto", "auto"},
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{OPUS_AUTO, "Auto", "auto"},
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{0, NULL, NULL}
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};
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static volatile GType id = 0;
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@ -286,12 +286,13 @@ gst_opus_enc_sink_setcaps (GstPad * pad, GstCaps * caps)
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enc->frame_samples = enc->sample_rate / 50;
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break;
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case 40:
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enc->frame_samples = enc->sample_rate / 20;
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enc->frame_samples = enc->sample_rate / 25;
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break;
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case 60:
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enc->frame_samples = 3 * enc->sample_rate / 50;
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break;
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default:
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GST_WARNING_OBJECT (enc, "Unsupported frame size: %d", enc->frame_size);
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return FALSE;
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break;
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}
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@ -664,62 +665,24 @@ gst_opus_enc_create_metadata_buffer (GstOpusEnc * enc)
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static gboolean
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gst_opus_enc_setup (GstOpusEnc * enc)
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{
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//gint error = OPUS_OK;
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int error = OPUS_OK;
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enc->setup = FALSE;
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#if 0
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#ifdef HAVE_OPUS_0_7
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enc->mode = opus_mode_create (enc->rate, enc->frame_size, &error);
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#else
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enc->mode =
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opus_mode_create (enc->rate, enc->n_channels, enc->frame_size, &error);
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#endif
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if (!enc->mode)
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goto mode_initialization_failed;
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#ifdef HAVE_OPUS_0_11
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opus_header_init (&enc->header, enc->mode, enc->frame_size, enc->n_channels);
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#else
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#ifdef HAVE_OPUS_0_7
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opus_header_init (&enc->header, enc->mode, enc->n_channels);
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#else
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opus_header_init (&enc->header, enc->mode);
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#endif
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#endif
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enc->header.nb_channels = enc->n_channels;
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#ifdef HAVE_OPUS_0_8
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enc->frame_size = enc->header.frame_size;
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#else
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opus_mode_info (enc->mode, OPUS_GET_FRAME_SIZE, &enc->frame_size);
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#endif
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#endif
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#if 0
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#ifdef HAVE_OPUS_0_11
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enc->state = opus_encoder_create_custom (enc->mode, enc->n_channels, &error);
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#else
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#ifdef HAVE_OPUS_0_7
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enc->state = opus_encoder_create (enc->mode, enc->n_channels, &error);
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#else
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enc->state = opus_encoder_create (enc->mode);
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#endif
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#endif
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#endif
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enc->state = opus_encoder_create (enc->sample_rate, enc->n_channels,
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enc->audio_or_voip ? OPUS_APPLICATION_AUDIO : OPUS_APPLICATION_VOIP);
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if (!enc->state)
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enc->audio_or_voip ? OPUS_APPLICATION_AUDIO : OPUS_APPLICATION_VOIP,
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&error);
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if (!enc->state || error != OPUS_OK)
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goto encoder_creation_failed;
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opus_encoder_ctl (enc->state, OPUS_SET_BITRATE (enc->bitrate), 0);
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opus_encoder_ctl (enc->state, OPUS_SET_BANDWIDTH (enc->bandwidth), 0);
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opus_encoder_ctl (enc->state, OPUS_SET_VBR_FLAG (!enc->cbr), 0);
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opus_encoder_ctl (enc->state, OPUS_SET_VBR (!enc->cbr), 0);
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opus_encoder_ctl (enc->state, OPUS_SET_VBR_CONSTRAINT (enc->constrained_vbr),
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0);
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opus_encoder_ctl (enc->state, OPUS_SET_COMPLEXITY (enc->complexity), 0);
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opus_encoder_ctl (enc->state, OPUS_SET_INBAND_FEC_FLAG (enc->inband_fec), 0);
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opus_encoder_ctl (enc->state, OPUS_SET_DTX_FLAG (enc->dtx), 0);
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opus_encoder_ctl (enc->state, OPUS_SET_INBAND_FEC (enc->inband_fec), 0);
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opus_encoder_ctl (enc->state, OPUS_SET_DTX (enc->dtx), 0);
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opus_encoder_ctl (enc->state,
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OPUS_SET_PACKET_LOSS_PERC (enc->packet_loss_percentage), 0);
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@ -879,6 +842,12 @@ gst_opus_enc_encode (GstOpusEnc * enc, gboolean flush)
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GST_ERROR_OBJECT (enc, "Encoding failed: %d", outsize);
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ret = GST_FLOW_ERROR;
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goto done;
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} else if (outsize != bytes_per_packet) {
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GST_WARNING_OBJECT (enc,
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"Encoded size %d is different from %d bytes per packet", outsize,
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bytes_per_packet);
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ret = GST_FLOW_ERROR;
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goto done;
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}
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GST_BUFFER_TIMESTAMP (outbuf) = enc->start_ts +
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@ -916,29 +885,10 @@ gst_opus_enc_chain (GstPad * pad, GstBuffer * buf)
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if (!enc->setup)
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goto not_setup;
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#if 0
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if (!enc->header_sent) {
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/* Opus streams begin with two headers; the initial header (with
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most of the codec setup parameters) which is mandated by the Ogg
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bitstream spec. The second header holds any comment fields.
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We merely need to make the headers, then pass them to libopus
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one at a time; libopus handles the additional Ogg bitstream
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constraints */
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GstBuffer *buf1, *buf2;
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GstCaps *caps;
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guchar data[100];
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/* create header buffer */
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opus_header_to_packet (&enc->header, data, 100);
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buf1 = gst_opus_enc_buffer_from_data (enc, data, 100, 0);
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/* create comment buffer */
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buf2 = gst_opus_enc_create_metadata_buffer (enc);
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/* mark and put on caps */
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caps = gst_pad_get_caps (enc->srcpad);
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caps = gst_opus_enc_set_header_on_caps (caps, buf1, buf2);
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gst_caps_set_simple (caps,
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"rate", G_TYPE_INT, enc->sample_rate,
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"channels", G_TYPE_INT, enc->n_channels,
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@ -950,26 +900,8 @@ gst_opus_enc_chain (GstPad * pad, GstBuffer * buf)
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enc->sample_rate, enc->n_channels, enc->frame_size);
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gst_pad_set_caps (enc->srcpad, caps);
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gst_buffer_set_caps (buf1, caps);
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gst_buffer_set_caps (buf2, caps);
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gst_caps_unref (caps);
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/* push out buffers */
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ret = gst_opus_enc_push_buffer (enc, buf1);
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if (ret != GST_FLOW_OK) {
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gst_buffer_unref (buf2);
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goto done;
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}
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ret = gst_opus_enc_push_buffer (enc, buf2);
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if (ret != GST_FLOW_OK)
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goto done;
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enc->header_sent = TRUE;
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}
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#endif
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GST_DEBUG_OBJECT (enc, "received buffer of %u bytes", GST_BUFFER_SIZE (buf));
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