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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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qtmux: Error out on discontinuities/gaps when muxing raw audio
When muxing raw audio, we have no way of storing timestamps but are just storing a continuous stream of audio samples. If the difference between the expected and the real timestamp becomes to big, we should error out instead of silently creating files with wrong A/V sync. https://bugzilla.gnome.org/show_bug.cgi?id=780679
This commit is contained in:
Sebastian Dröge
parent
e4cbefcb6c
commit
f0163a016c
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@ -146,6 +146,10 @@
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GST_DEBUG_CATEGORY_STATIC (gst_qt_mux_debug);
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#define GST_CAT_DEFAULT gst_qt_mux_debug
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#ifndef ABSDIFF
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#define ABSDIFF(a, b) ((a) > (b) ? (a) - (b) : (b) - (a))
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#endif
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/* Hacker notes.
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*
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* The basic building blocks of MP4 files are:
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@ -270,6 +274,7 @@ enum
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PROP_DO_CTTS,
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PROP_INTERLEAVE_BYTES,
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PROP_INTERLEAVE_TIME,
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PROP_MAX_RAW_AUDIO_DRIFT,
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};
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/* some spare for header size as well */
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@ -291,6 +296,7 @@ enum
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#define DEFAULT_RESERVED_BYTES_PER_SEC_PER_TRAK 550
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#define DEFAULT_INTERLEAVE_BYTES 0
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#define DEFAULT_INTERLEAVE_TIME 250*GST_MSECOND
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#define DEFAULT_MAX_RAW_AUDIO_DRIFT 40 * GST_MSECOND
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static void gst_qt_mux_finalize (GObject * object);
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@ -493,6 +499,11 @@ gst_qt_mux_class_init (GstQTMuxClass * klass)
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"Interleave between streams in nanoseconds",
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0, G_MAXUINT64, DEFAULT_INTERLEAVE_TIME,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_MAX_RAW_AUDIO_DRIFT,
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g_param_spec_uint64 ("max-raw-audio-drift", "Max Raw Audio Drift",
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"Maximum allowed drift of raw audio samples vs. timestamps in nanoseconds",
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0, G_MAXUINT64, DEFAULT_MAX_RAW_AUDIO_DRIFT,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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gstelement_class->request_new_pad =
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GST_DEBUG_FUNCPTR (gst_qt_mux_request_new_pad);
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@ -508,6 +519,7 @@ gst_qt_mux_pad_reset (GstQTPad * qtpad)
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qtpad->sample_size = 0;
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qtpad->sync = FALSE;
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qtpad->last_dts = 0;
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qtpad->sample_offset = 0;
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qtpad->dts_adjustment = GST_CLOCK_TIME_NONE;
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qtpad->first_ts = GST_CLOCK_TIME_NONE;
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qtpad->first_dts = GST_CLOCK_TIME_NONE;
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@ -655,6 +667,7 @@ gst_qt_mux_init (GstQTMux * qtmux, GstQTMuxClass * qtmux_klass)
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DEFAULT_RESERVED_BYTES_PER_SEC_PER_TRAK;
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qtmux->interleave_bytes = DEFAULT_INTERLEAVE_BYTES;
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qtmux->interleave_time = DEFAULT_INTERLEAVE_TIME;
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qtmux->max_raw_audio_drift = DEFAULT_MAX_RAW_AUDIO_DRIFT;
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/* always need this */
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qtmux->context =
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@ -3318,14 +3331,27 @@ gst_qt_mux_add_buffer (GstQTMux * qtmux, GstQTPad * pad, GstBuffer * buf)
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/* fragments only deal with 1 buffer == 1 chunk (== 1 sample) */
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if (pad->sample_size && !qtmux->fragment_sequence) {
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GstClockTime expected_timestamp;
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/* Constant size packets: usually raw audio (with many samples per
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buffer (= chunk)), but can also be fixed-packet-size codecs like ADPCM
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*/
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sample_size = pad->sample_size;
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if (gst_buffer_get_size (last_buf) % sample_size != 0)
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goto fragmented_sample;
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/* note: qt raw audio storage warps it implicitly into a timewise
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* perfect stream, discarding buffer times */
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* perfect stream, discarding buffer times.
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* If the difference between the current PTS and the expected one
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* becomes too big, we error out: there was a gap and we have no way to
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* represent that, causing A/V sync to be off */
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expected_timestamp =
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gst_util_uint64_scale (pad->sample_offset, GST_SECOND,
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atom_trak_get_timescale (pad->trak)) + pad->first_ts;
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if (ABSDIFF (GST_BUFFER_DTS_OR_PTS (last_buf),
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expected_timestamp) > qtmux->max_raw_audio_drift)
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goto raw_audio_timestamp_drift;
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if (GST_BUFFER_DURATION (last_buf) != GST_CLOCK_TIME_NONE) {
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nsamples = gst_util_uint64_scale_round (GST_BUFFER_DURATION (last_buf),
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atom_trak_get_timescale (pad->trak), GST_SECOND);
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@ -3339,7 +3365,9 @@ gst_qt_mux_add_buffer (GstQTMux * qtmux, GstQTPad * pad, GstBuffer * buf)
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/* timescale = samplerate */
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scaled_duration = 1;
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pad->last_dts += duration;
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pad->last_dts =
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pad->first_dts + gst_util_uint64_scale_round (pad->sample_offset +
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nsamples, GST_SECOND, atom_trak_get_timescale (pad->trak));
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} else {
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nsamples = 1;
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sample_size = gst_buffer_get_size (last_buf);
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@ -3371,6 +3399,8 @@ gst_qt_mux_add_buffer (GstQTMux * qtmux, GstQTPad * pad, GstBuffer * buf)
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}
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}
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pad->sample_offset += nsamples;
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/* for computing the avg bitrate */
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pad->total_bytes += gst_buffer_get_size (last_buf);
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pad->total_duration += duration;
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@ -3473,6 +3503,18 @@ fragmented_sample:
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("Audio buffer contains fragmented sample."));
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goto bail;
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}
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raw_audio_timestamp_drift:
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{
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/* TODO: Could in theory be implemented with edit lists */
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GST_ELEMENT_ERROR (qtmux, STREAM, MUX, (NULL),
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("Audio stream timestamps are drifting (got %" GST_TIME_FORMAT
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", expected %" GST_TIME_FORMAT "). This is not supported yet!",
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GST_TIME_ARGS (GST_BUFFER_DTS_OR_PTS (last_buf)),
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GST_TIME_ARGS (gst_util_uint64_scale (pad->sample_offset,
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GST_SECOND,
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atom_trak_get_timescale (pad->trak)) + pad->first_ts)));
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goto bail;
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}
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no_pts:
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{
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GST_ELEMENT_ERROR (qtmux, STREAM, MUX, (NULL), ("Buffer has no PTS."));
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@ -4923,6 +4965,9 @@ gst_qt_mux_get_property (GObject * object,
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case PROP_INTERLEAVE_TIME:
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g_value_set_uint64 (value, qtmux->interleave_time);
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break;
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case PROP_MAX_RAW_AUDIO_DRIFT:
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g_value_set_uint64 (value, qtmux->max_raw_audio_drift);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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@ -5008,6 +5053,9 @@ gst_qt_mux_set_property (GObject * object,
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qtmux->interleave_time = g_value_get_uint64 (value);
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qtmux->interleave_time_set = TRUE;
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break;
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case PROP_MAX_RAW_AUDIO_DRIFT:
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qtmux->max_raw_audio_drift = g_value_get_uint64 (value);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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@ -108,6 +108,7 @@ struct _GstQTPad
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GstBuffer *last_buf;
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/* dts of last_buf */
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GstClockTime last_dts;
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guint64 sample_offset;
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/* This is compensate for CTTS */
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GstClockTime dts_adjustment;
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@ -254,6 +255,8 @@ struct _GstQTMux
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GstClockTime interleave_time;
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gboolean interleave_bytes_set, interleave_time_set;
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GstClockTime max_raw_audio_drift;
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/* Reserved minimum MOOV size in bytes
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* This is converted from reserved_max_duration
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* using the bytes/trak/sec estimate */
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