}
/**
- * gst_audio_structure_set_int:
- * @structure: a #GstStructure
- * @flag: a set of #GstAudioFieldFlag
+ * gst_audio_info_to_caps:
+ * @info: a #GstAudioInfo
*
- * Do not use anymore.
+ * Convert the values of @info into a #GstCaps.
*
- * Deprecated: use gst_structure_set()
+ * Returns: (transfer full): the new #GstCaps containing the
+ * info of @info.
*/
-#ifndef GST_REMOVE_DEPRECATED
-#ifdef GST_DISABLE_DEPRECATED
-typedef enum
+GstCaps *
+gst_audio_info_to_caps (GstAudioInfo * info)
{
- GST_AUDIO_FIELD_RATE = (1 << 0),
- GST_AUDIO_FIELD_CHANNELS = (1 << 1),
- GST_AUDIO_FIELD_ENDIANNESS = (1 << 2),
- GST_AUDIO_FIELD_WIDTH = (1 << 3),
- GST_AUDIO_FIELD_DEPTH = (1 << 4),
- GST_AUDIO_FIELD_SIGNED = (1 << 5),
-} GstAudioFieldFlag;
-void
-gst_audio_structure_set_int (GstStructure * structure, GstAudioFieldFlag flag);
-#endif /* GST_DISABLE_DEPRECATED */
+ GstCaps *caps;
+ const gchar *format;
-void
-gst_audio_structure_set_int (GstStructure * structure, GstAudioFieldFlag flag)
+ g_return_val_if_fail (info != NULL, NULL);
+ g_return_val_if_fail (info->finfo != NULL, NULL);
+ g_return_val_if_fail (info->finfo->format != GST_AUDIO_FORMAT_UNKNOWN, NULL);
+
+ format = gst_audio_format_to_string (info->finfo->format);
+ g_return_val_if_fail (format != NULL, NULL);
+
+ caps = gst_caps_new_simple ("audio/x-raw",
+ "format", G_TYPE_STRING, format,
+ "rate", G_TYPE_INT, info->rate,
+ "channels", G_TYPE_INT, info->channels, NULL);
+
+ if (info->channels > 2) {
+ GValue pos_val_arr = { 0 }
+ , pos_val_entry = {
+ 0};
+ gint i, max_pos;
+ GstStructure *str;
+
+ /* build gvaluearray from positions */
+ g_value_init (&pos_val_arr, GST_TYPE_ARRAY);
+ g_value_init (&pos_val_entry, GST_TYPE_AUDIO_CHANNEL_POSITION);
+ max_pos = MAX (info->channels, 64);
+ for (i = 0; i < max_pos; i++) {
+ g_value_set_enum (&pos_val_entry, info->position[i]);
+ gst_value_array_append_value (&pos_val_arr, &pos_val_entry);
+ }
+ g_value_unset (&pos_val_entry);
+
+ /* add to structure */
+ str = gst_caps_get_structure (caps, 0);
+ gst_structure_set_value (str, "channel-positions", &pos_val_arr);
+ g_value_unset (&pos_val_arr);
+ }
+
+ return caps;
+}
+
+/**
+ * gst_audio_format_convert:
+ * @info: a #GstAudioInfo
+ * @src_format: #GstFormat of the @src_value
+ * @src_value: value to convert
+ * @dest_format: #GstFormat of the @dest_value
+ * @dest_value: pointer to destination value
+ *
+ * Converts among various #GstFormat types. This function handles
+ * GST_FORMAT_BYTES, GST_FORMAT_TIME, and GST_FORMAT_DEFAULT. For
+ * raw audio, GST_FORMAT_DEFAULT corresponds to audio frames. This
+ * function can be used to handle pad queries of the type GST_QUERY_CONVERT.
+ *
+ * Returns: TRUE if the conversion was successful.
+ */
+gboolean
+gst_audio_info_convert (GstAudioInfo * info,
+ GstFormat src_fmt, gint64 src_val, GstFormat dest_fmt, gint64 * dest_val)
{
- /* was added here:
- * http://webcvs.freedesktop.org/gstreamer/gst-plugins-base/gst-libs/gst/audio/audio.c?r1=1.16&r2=1.17
- * but it is not used
- */
- if (flag & GST_AUDIO_FIELD_RATE)
- gst_structure_set (structure, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT,
- NULL);
- if (flag & GST_AUDIO_FIELD_CHANNELS)
- gst_structure_set (structure, "channels", GST_TYPE_INT_RANGE, 1, G_MAXINT,
- NULL);
- if (flag & GST_AUDIO_FIELD_ENDIANNESS)
- _gst_audio_structure_set_list (structure, "endianness", G_TYPE_INT, 2,
- G_LITTLE_ENDIAN, G_BIG_ENDIAN, NULL);
- if (flag & GST_AUDIO_FIELD_WIDTH)
- _gst_audio_structure_set_list (structure, "width", G_TYPE_INT, 3, 8, 16, 32,
- NULL);
- if (flag & GST_AUDIO_FIELD_DEPTH)
- gst_structure_set (structure, "depth", GST_TYPE_INT_RANGE, 1, 32, NULL);
- if (flag & GST_AUDIO_FIELD_SIGNED)
- _gst_audio_structure_set_list (structure, "signed", G_TYPE_BOOLEAN, 2, TRUE,
- FALSE, NULL);
+ gboolean res = TRUE;
+ gint bpf, rate;
+
+ GST_DEBUG ("converting value %" G_GINT64_FORMAT " from %s (%d) to %s (%d)",
+ src_val, gst_format_get_name (src_fmt), src_fmt,
+ gst_format_get_name (dest_fmt), dest_fmt);
+
+ if (src_fmt == dest_fmt || src_val == -1) {
+ *dest_val = src_val;
+ goto done;
+ }
+
+ /* get important info */
+ bpf = GST_AUDIO_INFO_BPF (info);
+ rate = GST_AUDIO_INFO_RATE (info);
+
+ if (bpf == 0 || rate == 0) {
+ GST_DEBUG ("no rate or bpf configured");
+ res = FALSE;
+ goto done;
+ }
+
+ switch (src_fmt) {
+ case GST_FORMAT_BYTES:
+ switch (dest_fmt) {
+ case GST_FORMAT_TIME:
+ *dest_val = GST_FRAMES_TO_CLOCK_TIME (src_val / bpf, rate);
+ break;
+ case GST_FORMAT_DEFAULT:
+ *dest_val = src_val / bpf;
+ break;
+ default:
+ res = FALSE;
+ break;
+ }
+ break;
+ case GST_FORMAT_DEFAULT:
+ switch (dest_fmt) {
+ case GST_FORMAT_TIME:
+ *dest_val = GST_FRAMES_TO_CLOCK_TIME (src_val, rate);
+ break;
+ case GST_FORMAT_BYTES:
+ *dest_val = src_val * bpf;
+ break;
+ default:
+ res = FALSE;
+ break;
+ }
+ break;
+ case GST_FORMAT_TIME:
+ switch (dest_fmt) {
+ case GST_FORMAT_DEFAULT:
+ *dest_val = GST_CLOCK_TIME_TO_FRAMES (src_val, rate);
+ break;
+ case GST_FORMAT_BYTES:
+ *dest_val = GST_CLOCK_TIME_TO_FRAMES (src_val, rate);
+ *dest_val *= bpf;
+ break;
+ default:
+ res = FALSE;
+ break;
+ }
+ break;
+ default:
+ res = FALSE;
+ break;
+ }
+done:
+ GST_DEBUG ("ret=%d result %" G_GINT64_FORMAT, res, *dest_val);
+
+ return res;
}
-#endif /* GST_REMOVE_DEPRECATED */
+
+ #define SINT (GST_AUDIO_FORMAT_FLAG_INTEGER | GST_AUDIO_FORMAT_FLAG_SIGNED)
+ #define UINT (GST_AUDIO_FORMAT_FLAG_INTEGER)
+
+ #define MAKE_FORMAT(str,flags,end,width,depth,silent) \
+ { GST_AUDIO_FORMAT_ ##str, G_STRINGIFY(str), flags, end, width, depth, silent }
+
+ #define SILENT_0 { 0, 0, 0, 0, 0, 0, 0, 0 }
+ #define SILENT_U8 { 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80 }
+ #define SILENT_U16_LE { 0x00, 0x80, 0x00, 0x80, 0x00, 0x80, 0x00, 0x80 }
+ #define SILENT_U16_BE { 0x80, 0x00, 0x80, 0x00, 0x80, 0x00, 0x80, 0x00 }
+ #define SILENT_U24_LE { 0x00, 0x00, 0x80, 0x00, 0x00, 0x00, 0x80, 0x00 }
+ #define SILENT_U24_BE { 0x00, 0x80, 0x00, 0x00, 0x00, 0x80, 0x00, 0x00 }
+ #define SILENT_U32_LE { 0x00, 0x00, 0x00, 0x80, 0x00, 0x00, 0x00, 0x80 }
+ #define SILENT_U32_BE { 0x80, 0x00, 0x00, 0x00, 0x80, 0x00, 0x00, 0x00 }
+ #define SILENT_U24_3LE { 0x00, 0x00, 0x80, 0x00, 0x00, 0x80 }
+ #define SILENT_U24_3BE { 0x80, 0x00, 0x00, 0x80, 0x00, 0x00 }
+ #define SILENT_U20_3LE { 0x00, 0x00, 0x08, 0x00, 0x00, 0x08 }
+ #define SILENT_U20_3BE { 0x08, 0x00, 0x00, 0x08, 0x00, 0x00 }
+ #define SILENT_U18_3LE { 0x00, 0x00, 0x02, 0x00, 0x00, 0x02 }
+ #define SILENT_U18_3BE { 0x02, 0x00, 0x00, 0x02, 0x00, 0x00 }
+
+ static GstAudioFormatInfo formats[] = {
+ {GST_AUDIO_FORMAT_UNKNOWN, "UNKNOWN", 0, 0, 0, 0},
+ /* 8 bit */
+ MAKE_FORMAT (S8, SINT, 0, 8, 8, SILENT_0),
+ MAKE_FORMAT (U8, UINT, 0, 8, 8, SILENT_U8),
+ /* 16 bit */
+ MAKE_FORMAT (S16_LE, SINT, G_LITTLE_ENDIAN, 16, 16, SILENT_0),
+ MAKE_FORMAT (S16_BE, SINT, G_BIG_ENDIAN, 16, 16, SILENT_0),
+ MAKE_FORMAT (U16_LE, UINT, G_LITTLE_ENDIAN, 16, 16, SILENT_U16_LE),
+ MAKE_FORMAT (U16_BE, UINT, G_BIG_ENDIAN, 16, 16, SILENT_U16_BE),
+ /* 24 bit in low 3 bytes of 32 bits */
+ MAKE_FORMAT (S24_LE, SINT, G_LITTLE_ENDIAN, 32, 24, SILENT_0),
+ MAKE_FORMAT (S24_BE, SINT, G_BIG_ENDIAN, 32, 24, SILENT_0),
+ MAKE_FORMAT (U24_LE, UINT, G_LITTLE_ENDIAN, 32, 24, SILENT_U24_LE),
+ MAKE_FORMAT (U24_BE, UINT, G_BIG_ENDIAN, 32, 24, SILENT_U24_BE),
+ /* 32 bit */
+ MAKE_FORMAT (S32_LE, SINT, G_LITTLE_ENDIAN, 32, 32, SILENT_0),
+ MAKE_FORMAT (S32_BE, SINT, G_BIG_ENDIAN, 32, 32, SILENT_0),
+ MAKE_FORMAT (U32_LE, UINT, G_LITTLE_ENDIAN, 32, 32, SILENT_U32_LE),
+ MAKE_FORMAT (U32_BE, UINT, G_BIG_ENDIAN, 32, 32, SILENT_U32_BE),
+ /* 24 bit in 3 bytes */
+ MAKE_FORMAT (S24_3LE, SINT, G_LITTLE_ENDIAN, 24, 24, SILENT_0),
+ MAKE_FORMAT (S24_3BE, SINT, G_BIG_ENDIAN, 24, 24, SILENT_0),
+ MAKE_FORMAT (U24_3LE, UINT, G_LITTLE_ENDIAN, 24, 24, SILENT_U24_3LE),
+ MAKE_FORMAT (U24_3BE, UINT, G_BIG_ENDIAN, 24, 24, SILENT_U24_3BE),
+ /* 20 bit in 3 bytes */
+ MAKE_FORMAT (S20_3LE, SINT, G_LITTLE_ENDIAN, 24, 20, SILENT_0),
+ MAKE_FORMAT (S20_3BE, SINT, G_BIG_ENDIAN, 24, 20, SILENT_0),
+ MAKE_FORMAT (U20_3LE, UINT, G_LITTLE_ENDIAN, 24, 20, SILENT_U20_3LE),
+ MAKE_FORMAT (U20_3BE, UINT, G_BIG_ENDIAN, 24, 20, SILENT_U20_3BE),
+ /* 18 bit in 3 bytes */
+ MAKE_FORMAT (S18_3LE, SINT, G_LITTLE_ENDIAN, 24, 18, SILENT_0),
+ MAKE_FORMAT (S18_3BE, SINT, G_BIG_ENDIAN, 24, 18, SILENT_0),
+ MAKE_FORMAT (U18_3LE, UINT, G_LITTLE_ENDIAN, 24, 18, SILENT_U18_3LE),
+ MAKE_FORMAT (U18_3BE, UINT, G_BIG_ENDIAN, 24, 18, SILENT_U18_3BE),
+ /* float */
+ MAKE_FORMAT (F32_LE, GST_AUDIO_FORMAT_FLAG_FLOAT, G_LITTLE_ENDIAN, 32, 32,
+ SILENT_0),
+ MAKE_FORMAT (F32_BE, GST_AUDIO_FORMAT_FLAG_FLOAT, G_BIG_ENDIAN, 32, 32,
+ SILENT_0),
+ MAKE_FORMAT (F64_LE, GST_AUDIO_FORMAT_FLAG_FLOAT, G_LITTLE_ENDIAN, 64, 64,
+ SILENT_0),
+ MAKE_FORMAT (F64_BE, GST_AUDIO_FORMAT_FLAG_FLOAT, G_BIG_ENDIAN, 64, 64,
+ SILENT_0)
+ };
+
+ static GstAudioFormat
+ gst_audio_format_from_caps_structure (const GstStructure * s)
+ {
+ gint endianness, width, depth;
+ guint i;
+
+ if (gst_structure_has_name (s, "audio/x-raw-int")) {
+ gboolean sign;
+
+ if (!gst_structure_get_boolean (s, "signed", &sign))
+ goto missing_field_signed;
+
+ if (!gst_structure_get_int (s, "endianness", &endianness))
+ goto missing_field_endianness;
+
+ if (!gst_structure_get_int (s, "width", &width))
+ goto missing_field_width;
+
+ if (!gst_structure_get_int (s, "depth", &depth))
+ goto missing_field_depth;
+
+ for (i = 0; i < G_N_ELEMENTS (formats); i++) {
+ if (GST_AUDIO_FORMAT_INFO_IS_INTEGER (&formats[i]) &&
+ sign == GST_AUDIO_FORMAT_INFO_IS_SIGNED (&formats[i]) &&
+ GST_AUDIO_FORMAT_INFO_ENDIANNESS (&formats[i]) == endianness &&
+ GST_AUDIO_FORMAT_INFO_WIDTH (&formats[i]) == width &&
+ GST_AUDIO_FORMAT_INFO_DEPTH (&formats[i]) == depth) {
+ return GST_AUDIO_FORMAT_INFO_FORMAT (&formats[i]);
+ }
+ }
+ } else if (gst_structure_has_name (s, "audio/x-raw-float")) {
+ /* fallbacks are for backwards compatibility (is this needed at all?) */
+ if (!gst_structure_get_int (s, "endianness", &endianness)) {
+ GST_WARNING ("float audio caps without endianness %" GST_PTR_FORMAT, s);
+ endianness = G_BYTE_ORDER;
+ }
+
+ if (!gst_structure_get_int (s, "width", &width)) {
+ GST_WARNING ("float audio caps without width %" GST_PTR_FORMAT, s);
+ width = 32;
+ }
+
+ for (i = 0; i < G_N_ELEMENTS (formats); i++) {
+ if (GST_AUDIO_FORMAT_INFO_IS_FLOAT (&formats[i]) &&
+ GST_AUDIO_FORMAT_INFO_ENDIANNESS (&formats[i]) == endianness &&
+ GST_AUDIO_FORMAT_INFO_WIDTH (&formats[i]) == width) {
+ return GST_AUDIO_FORMAT_INFO_FORMAT (&formats[i]);
+ }
+ }
+ }
+
+ /* no match */
+ return GST_AUDIO_FORMAT_UNKNOWN;
+
+ missing_field_signed:
+ {
+ GST_ERROR ("missing 'signed' field in audio caps %" GST_PTR_FORMAT, s);
+ return GST_AUDIO_FORMAT_UNKNOWN;
+ }
+ missing_field_endianness:
+ {
+ GST_ERROR ("missing 'endianness' field in audio caps %" GST_PTR_FORMAT, s);
+ return GST_AUDIO_FORMAT_UNKNOWN;
+ }
+ missing_field_depth:
+ {
+ GST_ERROR ("missing 'depth' field in audio caps %" GST_PTR_FORMAT, s);
+ return GST_AUDIO_FORMAT_UNKNOWN;
+ }
+ missing_field_width:
+ {
+ GST_ERROR ("missing 'width' field in audio caps %" GST_PTR_FORMAT, s);
+ return GST_AUDIO_FORMAT_UNKNOWN;
+ }
+ }
+
+ /* FIXME: remove these if we don't actually go for deep alloc positions */
+ void
+ gst_audio_info_init (GstAudioInfo * info)
+ {
+ memset (info, 0, sizeof (GstAudioInfo));
+ }
+
+ void
+ gst_audio_info_clear (GstAudioInfo * info)
+ {
+ memset (info, 0, sizeof (GstAudioInfo));
+ }
+
+ GstAudioInfo *
+ gst_audio_info_copy (GstAudioInfo * info)
+ {
+ return (GstAudioInfo *) g_slice_copy (sizeof (GstAudioInfo), info);
+ }
+
+ void
+ gst_audio_info_free (GstAudioInfo * info)
+ {
+ g_slice_free (GstAudioInfo, info);
+ }
+
+ static void
+ gst_audio_info_set_format (GstAudioInfo * info, GstAudioFormat format,
+ gint rate, gint channels)
+ {
+ const GstAudioFormatInfo *finfo;
+
+ g_return_if_fail (info != NULL);
+ g_return_if_fail (format != GST_AUDIO_FORMAT_UNKNOWN);
+
+ finfo = &formats[format];
+
+ info->flags = 0;
+ info->finfo = finfo;
+ info->rate = rate;
+ info->channels = channels;
+ info->bpf = (finfo->width * channels) / 8;
+ }
+
+ /* from multichannel.c */
+ void priv_gst_audio_info_fill_default_channel_positions (GstAudioInfo * info);
+
+ /**
+ * gst_audio_info_from_caps:
+ * @info: a #GstAudioInfo
+ * @caps: a #GstCaps
+ *
+ * Parse @caps and update @info.
+ *
+ * Returns: TRUE if @caps could be parsed
+ *
+ * Since: 0.10.36
+ */
+ gboolean
+ gst_audio_info_from_caps (GstAudioInfo * info, const GstCaps * caps)
+ {
+ GstStructure *str;
+ GstAudioFormat format;
+ gint rate, channels;
+ const GValue *pos_val_arr, *pos_val_entry;
+ gint i;
+
+ g_return_val_if_fail (info != NULL, FALSE);
+ g_return_val_if_fail (caps != NULL, FALSE);
+ g_return_val_if_fail (gst_caps_is_fixed (caps), FALSE);
+
+ GST_DEBUG ("parsing caps %" GST_PTR_FORMAT, caps);
+
+ str = gst_caps_get_structure (caps, 0);
+
+ format = gst_audio_format_from_caps_structure (str);
+ if (format == GST_AUDIO_FORMAT_UNKNOWN)
+ goto unknown_format;
+
+ if (!gst_structure_get_int (str, "rate", &rate))
+ goto no_rate;
+ if (!gst_structure_get_int (str, "channels", &channels))
+ goto no_channels;
+
+ gst_audio_info_set_format (info, format, rate, channels);
+
+ pos_val_arr = gst_structure_get_value (str, "channel-positions");
+ if (pos_val_arr) {
+ if (channels <= G_N_ELEMENTS (info->position)) {
+ for (i = 0; i < channels; i++) {
+ pos_val_entry = gst_value_array_get_value (pos_val_arr, i);
+ info->position[i] = g_value_get_enum (pos_val_entry);
+ }
+ } else {
+ /* for that many channels, the positions are always NONE */
+ for (i = 0; i < G_N_ELEMENTS (info->position); i++)
+ info->position[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
+ info->flags |= GST_AUDIO_FLAG_DEFAULT_POSITIONS;
+ }
+ } else {
+ info->flags |= GST_AUDIO_FLAG_DEFAULT_POSITIONS;
+ priv_gst_audio_info_fill_default_channel_positions (info);
+ }
+
+ return TRUE;
+
+ /* ERROR */
+ unknown_format:
+ {
+ GST_ERROR ("unknown format given");
+ return FALSE;
+ }
+ no_rate:
+ {
+ GST_ERROR ("no rate property given");
+ return FALSE;
+ }
+ no_channels:
+ {
+ GST_ERROR ("no channels property given");
+ return FALSE;
+ }
+ }
+
+ /**
+ * gst_audio_info_to_caps:
+ * @info: a #GstAudioInfo
+ *
+ * Convert the values of @info into a #GstCaps.
+ *
+ * Returns: (transfer full): the new #GstCaps containing the
+ * info of @info.
+ *
+ * Since: 0.10.36
+ */
+ GstCaps *
+ gst_audio_info_to_caps (GstAudioInfo * info)
+ {
+ GstCaps *caps;
+
+ g_return_val_if_fail (info != NULL, NULL);
+ g_return_val_if_fail (info->finfo != NULL, NULL);
+ g_return_val_if_fail (info->finfo->format != GST_AUDIO_FORMAT_UNKNOWN, NULL);
+
+ if (GST_AUDIO_FORMAT_INFO_IS_INTEGER (info->finfo)) {
+ caps = gst_caps_new_simple ("audio/x-raw-int",
+ "width", G_TYPE_INT, GST_AUDIO_INFO_WIDTH (info),
+ "depth", G_TYPE_INT, GST_AUDIO_INFO_DEPTH (info),
+ "endianness", G_TYPE_INT,
+ GST_AUDIO_FORMAT_INFO_ENDIANNESS (info->finfo), "signed",
+ G_TYPE_BOOLEAN, GST_AUDIO_FORMAT_INFO_IS_SIGNED (info->finfo), "rate",
+ G_TYPE_INT, GST_AUDIO_INFO_RATE (info), "channels", G_TYPE_INT,
+ GST_AUDIO_INFO_CHANNELS (info), NULL);
+ } else if (GST_AUDIO_FORMAT_INFO_IS_FLOAT (info->finfo)) {
+ caps = gst_caps_new_simple ("audio/x-raw-float",
+ "width", G_TYPE_INT, GST_AUDIO_INFO_WIDTH (info),
+ "endianness", G_TYPE_INT,
+ GST_AUDIO_FORMAT_INFO_ENDIANNESS (info->finfo), "rate", G_TYPE_INT,
+ GST_AUDIO_INFO_RATE (info), "channels", G_TYPE_INT,
+ GST_AUDIO_INFO_CHANNELS (info), NULL);
+ } else {
+ GST_ERROR ("unknown audio format, neither integer nor float");
+ return NULL;
+ }
+
+ if (info->channels > 2) {
+ GValue pos_val_arr = { 0 }
+ , pos_val_entry = {
+ 0};
+ GstStructure *str;
+ gint i;
+
+ /* build gvaluearray from positions */
+ g_value_init (&pos_val_arr, GST_TYPE_ARRAY);
+ g_value_init (&pos_val_entry, GST_TYPE_AUDIO_CHANNEL_POSITION);
+ for (i = 0; i < info->channels; i++) {
+ /* if we have many many channels, all positions are NONE */
+ if (info->channels <= 64)
+ g_value_set_enum (&pos_val_entry, info->position[i]);
+ else
+ g_value_set_enum (&pos_val_entry, GST_AUDIO_CHANNEL_POSITION_NONE);
+
+ gst_value_array_append_value (&pos_val_arr, &pos_val_entry);
+ }
+ g_value_unset (&pos_val_entry);
+
+ /* add to structure */
+ str = gst_caps_get_structure (caps, 0);
+ gst_structure_set_value (str, "channel-positions", &pos_val_arr);
+ g_value_unset (&pos_val_arr);
+ }
+
+ return caps;
+ }
+
+ /**
+ * gst_audio_format_convert:
+ * @info: a #GstAudioInfo
+ * @src_format: #GstFormat of the @src_value
+ * @src_value: value to convert
+ * @dest_format: #GstFormat of the @dest_value
+ * @dest_value: pointer to destination value
+ *
+ * Converts among various #GstFormat types. This function handles
+ * GST_FORMAT_BYTES, GST_FORMAT_TIME, and GST_FORMAT_DEFAULT. For
+ * raw audio, GST_FORMAT_DEFAULT corresponds to audio frames. This
+ * function can be used to handle pad queries of the type GST_QUERY_CONVERT.
+ *
+ * Returns: TRUE if the conversion was successful.
+ *
+ * Since: 0.10.36
+ */
+ gboolean
+ gst_audio_info_convert (GstAudioInfo * info,
+ GstFormat src_fmt, gint64 src_val, GstFormat dest_fmt, gint64 * dest_val)
+ {
+ gboolean res = TRUE;
+ gint bpf, rate;
+
+ GST_DEBUG ("converting value %" G_GINT64_FORMAT " from %s (%d) to %s (%d)",
+ src_val, gst_format_get_name (src_fmt), src_fmt,
+ gst_format_get_name (dest_fmt), dest_fmt);
+
+ if (src_fmt == dest_fmt || src_val == -1) {
+ *dest_val = src_val;
+ goto done;
+ }
+
+ /* get important info */
+ bpf = GST_AUDIO_INFO_BPF (info);
+ rate = GST_AUDIO_INFO_RATE (info);
+
+ if (bpf == 0 || rate == 0) {
+ GST_DEBUG ("no rate or bpf configured");
+ res = FALSE;
+ goto done;
+ }
+
+ switch (src_fmt) {
+ case GST_FORMAT_BYTES:
+ switch (dest_fmt) {
+ case GST_FORMAT_TIME:
+ *dest_val = GST_FRAMES_TO_CLOCK_TIME (src_val / bpf, rate);
+ break;
+ case GST_FORMAT_DEFAULT:
+ *dest_val = src_val / bpf;
+ break;
+ default:
+ res = FALSE;
+ break;
+ }
+ break;
+ case GST_FORMAT_DEFAULT:
+ switch (dest_fmt) {
+ case GST_FORMAT_TIME:
+ *dest_val = GST_FRAMES_TO_CLOCK_TIME (src_val, rate);
+ break;
+ case GST_FORMAT_BYTES:
+ *dest_val = src_val * bpf;
+ break;
+ default:
+ res = FALSE;
+ break;
+ }
+ break;
+ case GST_FORMAT_TIME:
+ switch (dest_fmt) {
+ case GST_FORMAT_DEFAULT:
+ *dest_val = GST_CLOCK_TIME_TO_FRAMES (src_val, rate);
+ break;
+ case GST_FORMAT_BYTES:
+ *dest_val = GST_CLOCK_TIME_TO_FRAMES (src_val, rate);
+ *dest_val *= bpf;
+ break;
+ default:
+ res = FALSE;
+ break;
+ }
+ break;
+ default:
+ res = FALSE;
+ break;
+ }
+ done:
+ GST_DEBUG ("ret=%d result %" G_GINT64_FORMAT, res, *dest_val);
+
+ return res;
+ }
+
/**
* gst_audio_buffer_clip:
* @buffer: The buffer to clip.
--- /dev/null
- dec->priv->agg = ! !res;
+ /* GStreamer
+ * Copyright (C) 2009 Igalia S.L.
+ * Author: Iago Toral Quiroga <itoral@igalia.com>
+ * Copyright (C) 2011 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>.
+ * Copyright (C) 2011 Nokia Corporation. All rights reserved.
+ * Contact: Stefan Kost <stefan.kost@nokia.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+ /**
+ * SECTION:gstbaseaudiodecoder
+ * @short_description: Base class for audio decoders
+ * @see_also: #GstBaseTransform
+ * @since: 0.10.36
+ *
+ * This base class is for audio decoders turning encoded data into
+ * raw audio samples.
+ *
+ * GstBaseAudioDecoder and subclass should cooperate as follows.
+ * <orderedlist>
+ * <listitem>
+ * <itemizedlist><title>Configuration</title>
+ * <listitem><para>
+ * Initially, GstBaseAudioDecoder calls @start when the decoder element
+ * is activated, which allows subclass to perform any global setup.
+ * Base class (context) parameters can already be set according to subclass
+ * capabilities (or possibly upon receive more information in subsequent
+ * @set_format).
+ * </para></listitem>
+ * <listitem><para>
+ * GstBaseAudioDecoder calls @set_format to inform subclass of the format
+ * of input audio data that it is about to receive.
+ * While unlikely, it might be called more than once, if changing input
+ * parameters require reconfiguration.
+ * </para></listitem>
+ * <listitem><para>
+ * GstBaseAudioDecoder calls @stop at end of all processing.
+ * </para></listitem>
+ * </itemizedlist>
+ * </listitem>
+ * As of configuration stage, and throughout processing, GstBaseAudioDecoder
+ * provides various (context) parameters, e.g. describing the format of
+ * output audio data (valid when output caps have been caps) or current parsing state.
+ * Conversely, subclass can and should configure context to inform
+ * base class of its expectation w.r.t. buffer handling.
+ * <listitem>
+ * <itemizedlist>
+ * <title>Data processing</title>
+ * <listitem><para>
+ * Base class gathers input data, and optionally allows subclass
+ * to parse this into subsequently manageable (as defined by subclass)
+ * chunks. Such chunks are subsequently referred to as 'frames',
+ * though they may or may not correspond to 1 (or more) audio format frame.
+ * </para></listitem>
+ * <listitem><para>
+ * Input frame is provided to subclass' @handle_frame.
+ * </para></listitem>
+ * <listitem><para>
+ * If codec processing results in decoded data, subclass should call
+ * @gst_base_audio_decoder_finish_frame to have decoded data pushed
+ * downstream.
+ * </para></listitem>
+ * <listitem><para>
+ * Just prior to actually pushing a buffer downstream,
+ * it is passed to @pre_push. Subclass should either use this callback
+ * to arrange for additional downstream pushing or otherwise ensure such
+ * custom pushing occurs after at least a method call has finished since
+ * setting src pad caps.
+ * </para></listitem>
+ * <listitem><para>
+ * During the parsing process GstBaseAudioDecoderClass will handle both
+ * srcpad and sinkpad events. Sink events will be passed to subclass
+ * if @event callback has been provided.
+ * </para></listitem>
+ * </itemizedlist>
+ * </listitem>
+ * <listitem>
+ * <itemizedlist><title>Shutdown phase</title>
+ * <listitem><para>
+ * GstBaseAudioDecoder class calls @stop to inform the subclass that data
+ * parsing will be stopped.
+ * </para></listitem>
+ * </itemizedlist>
+ * </listitem>
+ * </orderedlist>
+ *
+ * Subclass is responsible for providing pad template caps for
+ * source and sink pads. The pads need to be named "sink" and "src". It also
+ * needs to set the fixed caps on srcpad, when the format is ensured. This
+ * is typically when base class calls subclass' @set_format function, though
+ * it might be delayed until calling @gst_base_audio_decoder_finish_frame.
+ *
+ * In summary, above process should have subclass concentrating on
+ * codec data processing while leaving other matters to base class,
+ * such as most notably timestamp handling. While it may exert more control
+ * in this area (see e.g. @pre_push), it is very much not recommended.
+ *
+ * In particular, base class will try to arrange for perfect output timestamps
+ * as much as possible while tracking upstream timestamps.
+ * To this end, if deviation between the next ideal expected perfect timestamp
+ * and upstream exceeds #GstBaseAudioDecoder:tolerance, then resync to upstream
+ * occurs (which would happen always if the tolerance mechanism is disabled).
+ *
+ * In non-live pipelines, baseclass can also (configurably) arrange for
+ * output buffer aggregation which may help to redue large(r) numbers of
+ * small(er) buffers being pushed and processed downstream.
+ *
+ * On the other hand, it should be noted that baseclass only provides limited
+ * seeking support (upon explicit subclass request), as full-fledged support
+ * should rather be left to upstream demuxer, parser or alike. This simple
+ * approach caters for seeking and duration reporting using estimated input
+ * bitrates.
+ *
+ * Things that subclass need to take care of:
+ * <itemizedlist>
+ * <listitem><para>Provide pad templates</para></listitem>
+ * <listitem><para>
+ * Set source pad caps when appropriate
+ * </para></listitem>
+ * <listitem><para>
+ * Set user-configurable properties to sane defaults for format and
+ * implementing codec at hand, and convey some subclass capabilities and
+ * expectations in context.
+ * </para></listitem>
+ * <listitem><para>
+ * Accept data in @handle_frame and provide encoded results to
+ * @gst_base_audio_decoder_finish_frame. If it is prepared to perform
+ * PLC, it should also accept NULL data in @handle_frame and provide for
+ * data for indicated duration.
+ * </para></listitem>
+ * </itemizedlist>
+ */
+
+ #ifdef HAVE_CONFIG_H
+ #include "config.h"
+ #endif
+
+ #define GST_USE_UNSTABLE_API
+ #include "gstbaseaudiodecoder.h"
+ #include <gst/pbutils/descriptions.h>
+
+ #include <string.h>
+
+ GST_DEBUG_CATEGORY (baseaudiodecoder_debug);
+ #define GST_CAT_DEFAULT baseaudiodecoder_debug
+
+ #define GST_BASE_AUDIO_DECODER_GET_PRIVATE(obj) \
+ (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_BASE_AUDIO_DECODER, \
+ GstBaseAudioDecoderPrivate))
+
+ enum
+ {
+ LAST_SIGNAL
+ };
+
+ enum
+ {
+ PROP_0,
+ PROP_LATENCY,
+ PROP_TOLERANCE,
+ PROP_PLC
+ };
+
+ #define DEFAULT_LATENCY 0
+ #define DEFAULT_TOLERANCE 0
+ #define DEFAULT_PLC FALSE
+
+ typedef struct _GstBaseAudioDecoderContext
+ {
+ /* input */
+ /* (output) audio format */
+ GstAudioInfo info;
+
+ /* parsing state */
+ gboolean eos;
+ gboolean sync;
+
+ /* misc */
+ gint delay;
+
+ /* output */
+ gboolean do_plc;
+ gboolean do_byte_time;
+ gint max_errors;
+ /* MT-protected (with LOCK) */
+ GstClockTime min_latency;
+ GstClockTime max_latency;
+ } GstBaseAudioDecoderContext;
+
+ struct _GstBaseAudioDecoderPrivate
+ {
+ /* activation status */
+ gboolean active;
+
+ /* input base/first ts as basis for output ts */
+ GstClockTime base_ts;
+ /* input samples processed and sent downstream so far (w.r.t. base_ts) */
+ guint64 samples;
+
+ /* collected input data */
+ GstAdapter *adapter;
+ /* tracking input ts for changes */
+ GstClockTime prev_ts;
+ /* frames obtained from input */
+ GQueue frames;
+ /* collected output data */
+ GstAdapter *adapter_out;
+ /* ts and duration for output data collected above */
+ GstClockTime out_ts, out_dur;
+ /* mark outgoing discont */
+ gboolean discont;
+
+ /* subclass gave all it could already */
+ gboolean drained;
+ /* subclass currently being forcibly drained */
+ gboolean force;
+
+ /* input bps estimatation */
+ /* global in bytes seen */
+ guint64 bytes_in;
+ /* global samples sent out */
+ guint64 samples_out;
+ /* bytes flushed during parsing */
+ guint sync_flush;
+ /* error count */
+ gint error_count;
+ /* codec id tag */
+ GstTagList *taglist;
+
+ /* whether circumstances allow output aggregation */
+ gint agg;
+
+ /* reverse playback queues */
+ /* collect input */
+ GList *gather;
+ /* to-be-decoded */
+ GList *decode;
+ /* reversed output */
+ GList *queued;
+
+ /* context storage */
+ GstBaseAudioDecoderContext ctx;
+
+ /* properties */
+ GstClockTime latency;
+ GstClockTime tolerance;
+ gboolean plc;
+
+ };
+
+
+ static void gst_base_audio_decoder_finalize (GObject * object);
+ static void gst_base_audio_decoder_set_property (GObject * object,
+ guint prop_id, const GValue * value, GParamSpec * pspec);
+ static void gst_base_audio_decoder_get_property (GObject * object,
+ guint prop_id, GValue * value, GParamSpec * pspec);
+
+ static void gst_base_audio_decoder_clear_queues (GstBaseAudioDecoder * dec);
+ static GstFlowReturn gst_base_audio_decoder_chain_reverse (GstBaseAudioDecoder *
+ dec, GstBuffer * buf);
+
+ static GstStateChangeReturn gst_base_audio_decoder_change_state (GstElement *
+ element, GstStateChange transition);
+ static gboolean gst_base_audio_decoder_sink_event (GstPad * pad,
+ GstEvent * event);
+ static gboolean gst_base_audio_decoder_src_event (GstPad * pad,
+ GstEvent * event);
+ static gboolean gst_base_audio_decoder_sink_setcaps (GstPad * pad,
+ GstCaps * caps);
+ static gboolean gst_base_audio_decoder_src_setcaps (GstPad * pad,
+ GstCaps * caps);
+ static GstFlowReturn gst_base_audio_decoder_chain (GstPad * pad,
+ GstBuffer * buf);
+ static gboolean gst_base_audio_decoder_src_query (GstPad * pad,
+ GstQuery * query);
+ static gboolean gst_base_audio_decoder_sink_query (GstPad * pad,
+ GstQuery * query);
+ static const GstQueryType *gst_base_audio_decoder_get_query_types (GstPad *
+ pad);
+ static void gst_base_audio_decoder_reset (GstBaseAudioDecoder * dec,
+ gboolean full);
+
+
+ GST_BOILERPLATE (GstBaseAudioDecoder, gst_base_audio_decoder, GstElement,
+ GST_TYPE_ELEMENT);
+
+ static void
+ gst_base_audio_decoder_base_init (gpointer g_class)
+ {
+ }
+
+ static void
+ gst_base_audio_decoder_class_init (GstBaseAudioDecoderClass * klass)
+ {
+ GObjectClass *gobject_class;
+ GstElementClass *element_class;
+
+ gobject_class = G_OBJECT_CLASS (klass);
+ element_class = GST_ELEMENT_CLASS (klass);
+
+ parent_class = g_type_class_peek_parent (klass);
+
+ g_type_class_add_private (klass, sizeof (GstBaseAudioDecoderPrivate));
+
+ GST_DEBUG_CATEGORY_INIT (baseaudiodecoder_debug, "baseaudiodecoder", 0,
+ "baseaudiodecoder element");
+
+ gobject_class->set_property = gst_base_audio_decoder_set_property;
+ gobject_class->get_property = gst_base_audio_decoder_get_property;
+ gobject_class->finalize = gst_base_audio_decoder_finalize;
+
+ element_class->change_state = gst_base_audio_decoder_change_state;
+
+ /* Properties */
+ g_object_class_install_property (gobject_class, PROP_LATENCY,
+ g_param_spec_int64 ("min-latency", "Minimum Latency",
+ "Aggregate output data to a minimum of latency time (ns)",
+ 0, G_MAXINT64, DEFAULT_LATENCY,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_TOLERANCE,
+ g_param_spec_int64 ("tolerance", "Tolerance",
+ "Perfect ts while timestamp jitter/imperfection within tolerance (ns)",
+ 0, G_MAXINT64, DEFAULT_TOLERANCE,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_PLC,
+ g_param_spec_boolean ("plc", "Packet Loss Concealment",
+ "Perform packet loss concealment (if supported)",
+ DEFAULT_PLC, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ }
+
+ static void
+ gst_base_audio_decoder_init (GstBaseAudioDecoder * dec,
+ GstBaseAudioDecoderClass * klass)
+ {
+ GstPadTemplate *pad_template;
+
+ GST_DEBUG_OBJECT (dec, "gst_base_audio_decoder_init");
+
+ dec->priv = GST_BASE_AUDIO_DECODER_GET_PRIVATE (dec);
+
+ /* Setup sink pad */
+ pad_template =
+ gst_element_class_get_pad_template (GST_ELEMENT_CLASS (klass), "sink");
+ g_return_if_fail (pad_template != NULL);
+
+ dec->sinkpad = gst_pad_new_from_template (pad_template, "sink");
+ gst_pad_set_event_function (dec->sinkpad,
+ GST_DEBUG_FUNCPTR (gst_base_audio_decoder_sink_event));
+ gst_pad_set_setcaps_function (dec->sinkpad,
+ GST_DEBUG_FUNCPTR (gst_base_audio_decoder_sink_setcaps));
+ gst_pad_set_chain_function (dec->sinkpad,
+ GST_DEBUG_FUNCPTR (gst_base_audio_decoder_chain));
+ gst_pad_set_query_function (dec->sinkpad,
+ GST_DEBUG_FUNCPTR (gst_base_audio_decoder_sink_query));
+ gst_element_add_pad (GST_ELEMENT (dec), dec->sinkpad);
+ GST_DEBUG_OBJECT (dec, "sinkpad created");
+
+ /* Setup source pad */
+ pad_template =
+ gst_element_class_get_pad_template (GST_ELEMENT_CLASS (klass), "src");
+ g_return_if_fail (pad_template != NULL);
+
+ dec->srcpad = gst_pad_new_from_template (pad_template, "src");
+ gst_pad_set_setcaps_function (dec->srcpad,
+ GST_DEBUG_FUNCPTR (gst_base_audio_decoder_src_setcaps));
+ gst_pad_set_event_function (dec->srcpad,
+ GST_DEBUG_FUNCPTR (gst_base_audio_decoder_src_event));
+ gst_pad_set_query_function (dec->srcpad,
+ GST_DEBUG_FUNCPTR (gst_base_audio_decoder_src_query));
+ gst_pad_set_query_type_function (dec->srcpad,
+ GST_DEBUG_FUNCPTR (gst_base_audio_decoder_get_query_types));
+ gst_pad_use_fixed_caps (dec->srcpad);
+ gst_element_add_pad (GST_ELEMENT (dec), dec->srcpad);
+ GST_DEBUG_OBJECT (dec, "srcpad created");
+
+ dec->priv->adapter = gst_adapter_new ();
+ dec->priv->adapter_out = gst_adapter_new ();
+ g_queue_init (&dec->priv->frames);
+
+ /* property default */
+ dec->priv->latency = DEFAULT_LATENCY;
+ dec->priv->tolerance = DEFAULT_TOLERANCE;
+ dec->priv->plc = DEFAULT_PLC;
+
+ /* init state */
+ gst_base_audio_decoder_reset (dec, TRUE);
+ GST_DEBUG_OBJECT (dec, "init ok");
+ }
+
+ static void
+ gst_base_audio_decoder_reset (GstBaseAudioDecoder * dec, gboolean full)
+ {
+ GST_DEBUG_OBJECT (dec, "gst_base_audio_decoder_reset");
+
+ GST_OBJECT_LOCK (dec);
+
+ if (full) {
+ dec->priv->active = FALSE;
+ dec->priv->bytes_in = 0;
+ dec->priv->samples_out = 0;
+ dec->priv->agg = -1;
+ dec->priv->error_count = 0;
+ gst_base_audio_decoder_clear_queues (dec);
+
+ gst_audio_info_clear (&dec->priv->ctx.info);
+ memset (&dec->priv->ctx, 0, sizeof (dec->priv->ctx));
+
+ if (dec->priv->taglist) {
+ gst_tag_list_free (dec->priv->taglist);
+ dec->priv->taglist = NULL;
+ }
+
+ gst_segment_init (&dec->segment, GST_FORMAT_TIME);
+ }
+
+ g_queue_foreach (&dec->priv->frames, (GFunc) gst_buffer_unref, NULL);
+ g_queue_clear (&dec->priv->frames);
+ gst_adapter_clear (dec->priv->adapter);
+ gst_adapter_clear (dec->priv->adapter_out);
+ dec->priv->out_ts = GST_CLOCK_TIME_NONE;
+ dec->priv->out_dur = 0;
+ dec->priv->prev_ts = GST_CLOCK_TIME_NONE;
+ dec->priv->drained = TRUE;
+ dec->priv->base_ts = GST_CLOCK_TIME_NONE;
+ dec->priv->samples = 0;
+ dec->priv->discont = TRUE;
+ dec->priv->sync_flush = FALSE;
+
+ GST_OBJECT_UNLOCK (dec);
+ }
+
+ static void
+ gst_base_audio_decoder_finalize (GObject * object)
+ {
+ GstBaseAudioDecoder *dec;
+
+ g_return_if_fail (GST_IS_BASE_AUDIO_DECODER (object));
+ dec = GST_BASE_AUDIO_DECODER (object);
+
+ if (dec->priv->adapter) {
+ g_object_unref (dec->priv->adapter);
+ }
+ if (dec->priv->adapter_out) {
+ g_object_unref (dec->priv->adapter_out);
+ }
+
+ G_OBJECT_CLASS (parent_class)->finalize (object);
+ }
+
+ /* automagically perform sanity checking of src caps;
+ * also extracts output data format */
+ static gboolean
+ gst_base_audio_decoder_src_setcaps (GstPad * pad, GstCaps * caps)
+ {
+ GstBaseAudioDecoder *dec;
+ gboolean res = TRUE;
+ guint old_rate;
+
+ dec = GST_BASE_AUDIO_DECODER (gst_pad_get_parent (pad));
+
+ GST_DEBUG_OBJECT (dec, "setting src caps %" GST_PTR_FORMAT, caps);
+
+ /* parse caps here to check subclass;
+ * also makes us aware of output format */
+ if (!gst_caps_is_fixed (caps))
+ goto refuse_caps;
+
+ /* adjust ts tracking to new sample rate */
+ old_rate = GST_AUDIO_INFO_RATE (&dec->priv->ctx.info);
+ if (GST_CLOCK_TIME_IS_VALID (dec->priv->base_ts) && old_rate) {
+ dec->priv->base_ts +=
+ GST_FRAMES_TO_CLOCK_TIME (dec->priv->samples, old_rate);
+ dec->priv->samples = 0;
+ }
+
+ if (!gst_audio_info_from_caps (&dec->priv->ctx.info, caps))
+ goto refuse_caps;
+
+ gst_object_unref (dec);
+ return res;
+
+ /* ERRORS */
+ refuse_caps:
+ {
+ GST_WARNING_OBJECT (dec, "rejected caps %" GST_PTR_FORMAT, caps);
+ gst_object_unref (dec);
+ return res;
+ }
+ }
+
+ static gboolean
+ gst_base_audio_decoder_sink_setcaps (GstPad * pad, GstCaps * caps)
+ {
+ GstBaseAudioDecoder *dec;
+ GstBaseAudioDecoderClass *klass;
+ gboolean res = TRUE;
+
+ dec = GST_BASE_AUDIO_DECODER (gst_pad_get_parent (pad));
+ klass = GST_BASE_AUDIO_DECODER_GET_CLASS (dec);
+
+ GST_DEBUG_OBJECT (dec, "caps: %" GST_PTR_FORMAT, caps);
+
+ /* NOTE pbutils only needed here */
+ /* TODO maybe (only) upstream demuxer/parser etc should handle this ? */
+ if (dec->priv->taglist)
+ gst_tag_list_free (dec->priv->taglist);
+ dec->priv->taglist = gst_tag_list_new ();
+ gst_pb_utils_add_codec_description_to_tag_list (dec->priv->taglist,
+ GST_TAG_AUDIO_CODEC, caps);
+
+ if (klass->set_format)
+ res = klass->set_format (dec, caps);
+
+ g_object_unref (dec);
+ return res;
+ }
+
+ static void
+ gst_base_audio_decoder_setup (GstBaseAudioDecoder * dec)
+ {
+ GstQuery *query;
+ gboolean res;
+
+ /* check if in live pipeline, then latency messing is no-no */
+ query = gst_query_new_latency ();
+ res = gst_pad_peer_query (dec->sinkpad, query);
+ if (res) {
+ gst_query_parse_latency (query, &res, NULL, NULL);
+ res = !res;
+ }
+ gst_query_unref (query);
+
+ /* normalize to bool */
++ dec->priv->agg = !!res;
+ }
+
+ /* mini aggregator combining output buffers into fewer larger ones,
+ * if so allowed/configured */
+ static GstFlowReturn
+ gst_base_audio_decoder_output (GstBaseAudioDecoder * dec, GstBuffer * buf)
+ {
+ GstBaseAudioDecoderClass *klass;
+ GstBaseAudioDecoderPrivate *priv;
+ GstBaseAudioDecoderContext *ctx;
+ GstFlowReturn ret = GST_FLOW_OK;
+ GstBuffer *inbuf = NULL;
+
+ klass = GST_BASE_AUDIO_DECODER_GET_CLASS (dec);
+ priv = dec->priv;
+ ctx = &dec->priv->ctx;
+
+ if (G_UNLIKELY (priv->agg < 0))
+ gst_base_audio_decoder_setup (dec);
+
+ if (G_LIKELY (buf)) {
+ g_return_val_if_fail (ctx->info.bpf != 0, GST_FLOW_ERROR);
+
+ GST_LOG_OBJECT (dec, "output buffer of size %d with ts %" GST_TIME_FORMAT
+ ", duration %" GST_TIME_FORMAT, GST_BUFFER_SIZE (buf),
+ GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
+ GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
+
+ /* clip buffer */
+ buf = gst_audio_buffer_clip (buf, &dec->segment, ctx->info.rate,
+ ctx->info.bpf);
+ if (G_UNLIKELY (!buf)) {
+ GST_DEBUG_OBJECT (dec, "no data after clipping to segment");
+ } else {
+ GST_LOG_OBJECT (dec,
+ "buffer after segment clipping has size %d with ts %" GST_TIME_FORMAT
+ ", duration %" GST_TIME_FORMAT, GST_BUFFER_SIZE (buf),
+ GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
+ GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
+ }
+ } else {
+ GST_DEBUG_OBJECT (dec, "no output buffer");
+ }
+
+ again:
+ inbuf = NULL;
+ if (priv->agg && dec->priv->latency > 0) {
+ gint av;
+ gboolean assemble = FALSE;
+ const GstClockTimeDiff tol = 10 * GST_MSECOND;
+ GstClockTimeDiff diff = -100 * GST_MSECOND;
+
+ av = gst_adapter_available (priv->adapter_out);
+ if (G_UNLIKELY (!buf)) {
+ /* forcibly send current */
+ assemble = TRUE;
+ GST_LOG_OBJECT (dec, "forcing fragment flush");
+ } else if (av && (!GST_BUFFER_TIMESTAMP_IS_VALID (buf) ||
+ !GST_CLOCK_TIME_IS_VALID (priv->out_ts) ||
+ ((diff = GST_CLOCK_DIFF (GST_BUFFER_TIMESTAMP (buf),
+ priv->out_ts + priv->out_dur)) > tol) || diff < -tol)) {
+ assemble = TRUE;
+ GST_LOG_OBJECT (dec, "buffer %d ms apart from current fragment",
+ (gint) (diff / GST_MSECOND));
+ } else {
+ /* add or start collecting */
+ if (!av) {
+ GST_LOG_OBJECT (dec, "starting new fragment");
+ priv->out_ts = GST_BUFFER_TIMESTAMP (buf);
+ } else {
+ GST_LOG_OBJECT (dec, "adding to fragment");
+ }
+ gst_adapter_push (priv->adapter_out, buf);
+ priv->out_dur += GST_BUFFER_DURATION (buf);
+ av += GST_BUFFER_SIZE (buf);
+ buf = NULL;
+ }
+ if (priv->out_dur > dec->priv->latency)
+ assemble = TRUE;
+ if (av && assemble) {
+ GST_LOG_OBJECT (dec, "assembling fragment");
+ inbuf = buf;
+ buf = gst_adapter_take_buffer (priv->adapter_out, av);
+ GST_BUFFER_TIMESTAMP (buf) = priv->out_ts;
+ GST_BUFFER_DURATION (buf) = priv->out_dur;
+ priv->out_ts = GST_CLOCK_TIME_NONE;
+ priv->out_dur = 0;
+ }
+ }
+
+ if (G_LIKELY (buf)) {
+
+ /* decorate */
+ gst_buffer_set_caps (buf, GST_PAD_CAPS (dec->srcpad));
+
+ if (G_UNLIKELY (priv->discont)) {
+ GST_LOG_OBJECT (dec, "marking discont");
+ GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
+ priv->discont = FALSE;
+ }
+
+ if (G_LIKELY (GST_BUFFER_TIMESTAMP_IS_VALID (buf))) {
+ /* duration should always be valid for raw audio */
+ g_assert (GST_BUFFER_DURATION_IS_VALID (buf));
+ dec->segment.last_stop =
+ GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf);
+ }
+
+ if (klass->pre_push) {
+ /* last chance for subclass to do some dirty stuff */
+ ret = klass->pre_push (dec, &buf);
+ if (ret != GST_FLOW_OK || !buf) {
+ GST_DEBUG_OBJECT (dec, "subclass returned %s, buf %p",
+ gst_flow_get_name (ret), buf);
+ if (buf)
+ gst_buffer_unref (buf);
+ goto exit;
+ }
+ }
+
+ GST_LOG_OBJECT (dec, "pushing buffer of size %d with ts %" GST_TIME_FORMAT
+ ", duration %" GST_TIME_FORMAT, GST_BUFFER_SIZE (buf),
+ GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
+ GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
+
+ if (dec->segment.rate > 0.0) {
+ ret = gst_pad_push (dec->srcpad, buf);
+ GST_LOG_OBJECT (dec, "buffer pushed: %s", gst_flow_get_name (ret));
+ } else {
+ ret = GST_FLOW_OK;
+ priv->queued = g_list_prepend (priv->queued, buf);
+ GST_LOG_OBJECT (dec, "buffer queued");
+ }
+
+ exit:
+ if (inbuf) {
+ buf = inbuf;
+ goto again;
+ }
+ }
+
+ return ret;
+ }
+
+ GstFlowReturn
+ gst_base_audio_decoder_finish_frame (GstBaseAudioDecoder * dec, GstBuffer * buf,
+ gint frames)
+ {
+ GstBaseAudioDecoderPrivate *priv;
+ GstBaseAudioDecoderContext *ctx;
+ gint samples = 0;
+ GstClockTime ts, next_ts;
+
+ /* subclass should know what it is producing by now */
+ g_return_val_if_fail (buf == NULL || GST_PAD_CAPS (dec->srcpad) != NULL,
+ GST_FLOW_ERROR);
+ /* subclass should not hand us no data */
+ g_return_val_if_fail (buf == NULL || GST_BUFFER_SIZE (buf) > 0,
+ GST_FLOW_ERROR);
+ /* no dummy calls please */
+ g_return_val_if_fail (frames != 0, GST_FLOW_ERROR);
+
+ priv = dec->priv;
+ ctx = &dec->priv->ctx;
+
+ GST_LOG_OBJECT (dec, "accepting %d bytes == %d samples for %d frames",
+ buf ? GST_BUFFER_SIZE (buf) : -1,
+ buf ? GST_BUFFER_SIZE (buf) / ctx->info.bpf : -1, frames);
+
+ /* output shoud be whole number of sample frames */
+ if (G_LIKELY (buf && ctx->info.bpf)) {
+ if (GST_BUFFER_SIZE (buf) % ctx->info.bpf)
+ goto wrong_buffer;
+ /* per channel least */
+ samples = GST_BUFFER_SIZE (buf) / ctx->info.bpf;
+ }
+
+ /* frame and ts book-keeping */
+ if (G_UNLIKELY (frames < 0)) {
+ if (G_UNLIKELY (-frames - 1 > priv->frames.length))
+ goto overflow;
+ frames = priv->frames.length + frames + 1;
+ } else if (G_UNLIKELY (frames > priv->frames.length)) {
+ if (G_LIKELY (!priv->force)) {
+ /* no way we can let this pass */
+ g_assert_not_reached ();
+ /* really no way */
+ goto overflow;
+ }
+ }
+
+ if (G_LIKELY (priv->frames.length))
+ ts = GST_BUFFER_TIMESTAMP (priv->frames.head->data);
+ else
+ ts = GST_CLOCK_TIME_NONE;
+
+ GST_DEBUG_OBJECT (dec, "leading frame ts %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (ts));
+
+ while (priv->frames.length && frames) {
+ gst_buffer_unref (g_queue_pop_head (&priv->frames));
+ dec->priv->ctx.delay = dec->priv->frames.length;
+ frames--;
+ }
+
+ /* lock on */
+ if (G_UNLIKELY (!GST_CLOCK_TIME_IS_VALID (priv->base_ts))) {
+ priv->base_ts = ts;
+ GST_DEBUG_OBJECT (dec, "base_ts now %" GST_TIME_FORMAT, GST_TIME_ARGS (ts));
+ }
+
+ if (G_UNLIKELY (!buf))
+ goto exit;
+
+ /* slightly convoluted approach caters for perfect ts if subclass desires */
+ if (GST_CLOCK_TIME_IS_VALID (ts)) {
+ if (dec->priv->tolerance > 0) {
+ GstClockTimeDiff diff;
+
+ g_assert (GST_CLOCK_TIME_IS_VALID (priv->base_ts));
+ next_ts = priv->base_ts +
+ gst_util_uint64_scale (samples, GST_SECOND, ctx->info.rate);
+ GST_LOG_OBJECT (dec, "buffer is %d samples past base_ts %" GST_TIME_FORMAT
+ ", expected ts %" GST_TIME_FORMAT, samples,
+ GST_TIME_ARGS (priv->base_ts), GST_TIME_ARGS (next_ts));
+ diff = GST_CLOCK_DIFF (next_ts, ts);
+ GST_LOG_OBJECT (dec, "ts diff %d ms", (gint) (diff / GST_MSECOND));
+ /* if within tolerance,
+ * discard buffer ts and carry on producing perfect stream,
+ * otherwise resync to ts */
+ if (G_UNLIKELY (diff < -dec->priv->tolerance ||
+ diff > dec->priv->tolerance)) {
+ GST_DEBUG_OBJECT (dec, "base_ts resync");
+ priv->base_ts = ts;
+ priv->samples = 0;
+ }
+ } else {
+ GST_DEBUG_OBJECT (dec, "base_ts resync");
+ priv->base_ts = ts;
+ priv->samples = 0;
+ }
+ }
+
+ /* delayed one-shot stuff until confirmed data */
+ if (priv->taglist) {
+ GST_DEBUG_OBJECT (dec, "codec tag %" GST_PTR_FORMAT, priv->taglist);
+ if (gst_tag_list_is_empty (priv->taglist)) {
+ gst_tag_list_free (priv->taglist);
+ } else {
+ gst_element_found_tags (GST_ELEMENT (dec), priv->taglist);
+ }
+ priv->taglist = NULL;
+ }
+
+ buf = gst_buffer_make_metadata_writable (buf);
+ if (G_LIKELY (GST_CLOCK_TIME_IS_VALID (priv->base_ts))) {
+ GST_BUFFER_TIMESTAMP (buf) =
+ priv->base_ts +
+ GST_FRAMES_TO_CLOCK_TIME (priv->samples, ctx->info.rate);
+ GST_BUFFER_DURATION (buf) = priv->base_ts +
+ GST_FRAMES_TO_CLOCK_TIME (priv->samples + samples, ctx->info.rate) -
+ GST_BUFFER_TIMESTAMP (buf);
+ } else {
+ GST_BUFFER_TIMESTAMP (buf) = GST_CLOCK_TIME_NONE;
+ GST_BUFFER_DURATION (buf) =
+ GST_FRAMES_TO_CLOCK_TIME (samples, ctx->info.rate);
+ }
+ priv->samples += samples;
+ priv->samples_out += samples;
+
+ /* we got data, so note things are looking up */
+ if (G_UNLIKELY (dec->priv->error_count))
+ dec->priv->error_count--;
+
+ exit:
+ return gst_base_audio_decoder_output (dec, buf);
+
+ /* ERRORS */
+ wrong_buffer:
+ {
+ GST_ELEMENT_ERROR (dec, STREAM, ENCODE, (NULL),
+ ("buffer size %d not a multiple of %d", GST_BUFFER_SIZE (buf),
+ ctx->info.bpf));
+ gst_buffer_unref (buf);
+ return GST_FLOW_ERROR;
+ }
+ overflow:
+ {
+ GST_ELEMENT_ERROR (dec, STREAM, ENCODE,
+ ("received more decoded frames %d than provided %d", frames,
+ priv->frames.length), (NULL));
+ if (buf)
+ gst_buffer_unref (buf);
+ return GST_FLOW_ERROR;
+ }
+ }
+
+ static GstFlowReturn
+ gst_base_audio_decoder_handle_frame (GstBaseAudioDecoder * dec,
+ GstBaseAudioDecoderClass * klass, GstBuffer * buffer)
+ {
+ if (G_LIKELY (buffer)) {
+ /* keep around for admin */
+ GST_LOG_OBJECT (dec, "tracking frame size %d, ts %" GST_TIME_FORMAT,
+ GST_BUFFER_SIZE (buffer),
+ GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
+ g_queue_push_tail (&dec->priv->frames, buffer);
+ dec->priv->ctx.delay = dec->priv->frames.length;
+ dec->priv->bytes_in += GST_BUFFER_SIZE (buffer);
+ } else {
+ GST_LOG_OBJECT (dec, "providing subclass with NULL frame");
+ }
+
+ return klass->handle_frame (dec, buffer);
+ }
+
+ /* maybe subclass configurable instead, but this allows for a whole lot of
+ * raw samples, so at least quite some encoded ... */
+ #define GST_BASE_AUDIO_DECODER_MAX_SYNC 10 * 8 * 2 * 1024
+
+ static GstFlowReturn
+ gst_base_audio_decoder_push_buffers (GstBaseAudioDecoder * dec, gboolean force)
+ {
+ GstBaseAudioDecoderClass *klass;
+ GstBaseAudioDecoderPrivate *priv;
+ GstBaseAudioDecoderContext *ctx;
+ GstFlowReturn ret = GST_FLOW_OK;
+ GstBuffer *buffer;
+ gint av, flush;
+
+ klass = GST_BASE_AUDIO_DECODER_GET_CLASS (dec);
+ priv = dec->priv;
+ ctx = &dec->priv->ctx;
+
+ g_return_val_if_fail (klass->handle_frame != NULL, GST_FLOW_ERROR);
+
+ av = gst_adapter_available (priv->adapter);
+ GST_DEBUG_OBJECT (dec, "available: %d", av);
+
+ while (ret == GST_FLOW_OK) {
+
+ flush = 0;
+ ctx->eos = force;
+
+ if (G_LIKELY (av)) {
+ gint len;
+ GstClockTime ts;
+
+ /* parse if needed */
+ if (klass->parse) {
+ gint offset = 0;
+
+ /* limited (legacy) parsing; avoid whole of baseparse */
+ GST_DEBUG_OBJECT (dec, "parsing available: %d", av);
+ /* piggyback sync state on discont */
+ ctx->sync = !priv->discont;
+ ret = klass->parse (dec, priv->adapter, &offset, &len);
+
+ g_assert (offset <= av);
+ if (offset) {
+ /* jumped a bit */
+ GST_DEBUG_OBJECT (dec, "setting DISCONT");
+ gst_adapter_flush (priv->adapter, offset);
+ flush = offset;
+ /* avoid parsing indefinitely */
+ priv->sync_flush += offset;
+ if (priv->sync_flush > GST_BASE_AUDIO_DECODER_MAX_SYNC)
+ goto parse_failed;
+ }
+
+ if (ret == GST_FLOW_UNEXPECTED) {
+ GST_LOG_OBJECT (dec, "no frame yet");
+ ret = GST_FLOW_OK;
+ break;
+ } else if (ret == GST_FLOW_OK) {
+ GST_LOG_OBJECT (dec, "frame at offset %d of length %d", offset, len);
+ g_assert (offset + len <= av);
+ priv->sync_flush = 0;
+ } else {
+ break;
+ }
+ } else {
+ len = av;
+ }
+ /* track upstream ts, but do not get stuck if nothing new upstream */
+ ts = gst_adapter_prev_timestamp (priv->adapter, NULL);
+ if (ts == priv->prev_ts) {
+ GST_LOG_OBJECT (dec, "ts == prev_ts; discarding");
+ ts = GST_CLOCK_TIME_NONE;
+ } else {
+ priv->prev_ts = ts;
+ }
+ buffer = gst_adapter_take_buffer (priv->adapter, len);
+ buffer = gst_buffer_make_metadata_writable (buffer);
+ GST_BUFFER_TIMESTAMP (buffer) = ts;
+ flush += len;
+ } else {
+ if (!force)
+ break;
+ buffer = NULL;
+ }
+
+ ret = gst_base_audio_decoder_handle_frame (dec, klass, buffer);
+
+ /* do not keep pushing it ... */
+ if (G_UNLIKELY (!av)) {
+ priv->drained = TRUE;
+ break;
+ }
+
+ av -= flush;
+ g_assert (av >= 0);
+ }
+
+ GST_LOG_OBJECT (dec, "done pushing to subclass");
+ return ret;
+
+ /* ERRORS */
+ parse_failed:
+ {
+ GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("failed to parse stream"));
+ return GST_FLOW_ERROR;
+ }
+ }
+
+ static GstFlowReturn
+ gst_base_audio_decoder_drain (GstBaseAudioDecoder * dec)
+ {
+ GstFlowReturn ret;
+
+ if (dec->priv->drained)
+ return GST_FLOW_OK;
+ else {
+ /* dispatch reverse pending buffers */
+ /* chain eventually calls upon drain as well, but by that time
+ * gather list should be clear, so ok ... */
+ if (dec->segment.rate < 0.0 && dec->priv->gather)
+ gst_base_audio_decoder_chain_reverse (dec, NULL);
+ /* have subclass give all it can */
+ ret = gst_base_audio_decoder_push_buffers (dec, TRUE);
+ /* ensure all output sent */
+ ret = gst_base_audio_decoder_output (dec, NULL);
+ /* everything should be away now */
+ if (dec->priv->frames.length) {
+ /* not fatal/impossible though if subclass/codec eats stuff */
+ GST_WARNING_OBJECT (dec, "still %d frames left after draining",
+ dec->priv->frames.length);
+ g_queue_foreach (&dec->priv->frames, (GFunc) gst_buffer_unref, NULL);
+ g_queue_clear (&dec->priv->frames);
+ }
+ /* discard (unparsed) leftover */
+ gst_adapter_clear (dec->priv->adapter);
+
+ return ret;
+ }
+ }
+
+ /* hard == FLUSH, otherwise discont */
+ static GstFlowReturn
+ gst_base_audio_decoder_flush (GstBaseAudioDecoder * dec, gboolean hard)
+ {
+ GstBaseAudioDecoderClass *klass;
+ GstFlowReturn ret = GST_FLOW_OK;
+
+ klass = GST_BASE_AUDIO_DECODER_GET_CLASS (dec);
+
+ GST_LOG_OBJECT (dec, "flush hard %d", hard);
+
+ if (!hard) {
+ ret = gst_base_audio_decoder_drain (dec);
+ } else {
+ gst_base_audio_decoder_clear_queues (dec);
+ gst_segment_init (&dec->segment, GST_FORMAT_TIME);
+ dec->priv->error_count = 0;
+ }
+ /* only bother subclass with flushing if known it is already alive
+ * and kicking out stuff */
+ if (klass->flush && dec->priv->samples_out > 0)
+ klass->flush (dec, hard);
+ /* and get (re)set for the sequel */
+ gst_base_audio_decoder_reset (dec, FALSE);
+
+ return ret;
+ }
+
+ static GstFlowReturn
+ gst_base_audio_decoder_chain_forward (GstBaseAudioDecoder * dec,
+ GstBuffer * buffer)
+ {
+ GstFlowReturn ret;
+
+ /* grab buffer */
+ gst_adapter_push (dec->priv->adapter, buffer);
+ buffer = NULL;
+ /* new stuff, so we can push subclass again */
+ dec->priv->drained = FALSE;
+
+ /* hand to subclass */
+ ret = gst_base_audio_decoder_push_buffers (dec, FALSE);
+
+ GST_LOG_OBJECT (dec, "chain-done");
+ return ret;
+ }
+
+ static void
+ gst_base_audio_decoder_clear_queues (GstBaseAudioDecoder * dec)
+ {
+ GstBaseAudioDecoderPrivate *priv = dec->priv;
+
+ g_list_foreach (priv->queued, (GFunc) gst_mini_object_unref, NULL);
+ g_list_free (priv->queued);
+ priv->queued = NULL;
+ g_list_foreach (priv->gather, (GFunc) gst_mini_object_unref, NULL);
+ g_list_free (priv->gather);
+ priv->gather = NULL;
+ g_list_foreach (priv->decode, (GFunc) gst_mini_object_unref, NULL);
+ g_list_free (priv->decode);
+ priv->decode = NULL;
+ }
+
+ /*
+ * Input:
+ * Buffer decoding order: 7 8 9 4 5 6 3 1 2 EOS
+ * Discont flag: D D D D
+ *
+ * - Each Discont marks a discont in the decoding order.
+ *
+ * for vorbis, each buffer is a keyframe when we have the previous
+ * buffer. This means that to decode buffer 7, we need buffer 6, which
+ * arrives out of order.
+ *
+ * we first gather buffers in the gather queue until we get a DISCONT. We
+ * prepend each incomming buffer so that they are in reversed order.
+ *
+ * gather queue: 9 8 7
+ * decode queue:
+ * output queue:
+ *
+ * When a DISCONT is received (buffer 4), we move the gather queue to the
+ * decode queue. This is simply done be taking the head of the gather queue
+ * and prepending it to the decode queue. This yields:
+ *
+ * gather queue:
+ * decode queue: 7 8 9
+ * output queue:
+ *
+ * Then we decode each buffer in the decode queue in order and put the output
+ * buffer in the output queue. The first buffer (7) will not produce any output
+ * because it needs the previous buffer (6) which did not arrive yet. This
+ * yields:
+ *
+ * gather queue:
+ * decode queue: 7 8 9
+ * output queue: 9 8
+ *
+ * Then we remove the consumed buffers from the decode queue. Buffer 7 is not
+ * completely consumed, we need to keep it around for when we receive buffer
+ * 6. This yields:
+ *
+ * gather queue:
+ * decode queue: 7
+ * output queue: 9 8
+ *
+ * Then we accumulate more buffers:
+ *
+ * gather queue: 6 5 4
+ * decode queue: 7
+ * output queue:
+ *
+ * prepending to the decode queue on DISCONT yields:
+ *
+ * gather queue:
+ * decode queue: 4 5 6 7
+ * output queue:
+ *
+ * after decoding and keeping buffer 4:
+ *
+ * gather queue:
+ * decode queue: 4
+ * output queue: 7 6 5
+ *
+ * Etc..
+ */
+ static GstFlowReturn
+ gst_base_audio_decoder_flush_decode (GstBaseAudioDecoder * dec)
+ {
+ GstBaseAudioDecoderPrivate *priv = dec->priv;
+ GstFlowReturn res = GST_FLOW_OK;
+ GList *walk;
+
+ walk = priv->decode;
+
+ GST_DEBUG_OBJECT (dec, "flushing buffers to decoder");
+
+ /* clear buffer and decoder state */
+ gst_base_audio_decoder_flush (dec, FALSE);
+
+ while (walk) {
+ GList *next;
+ GstBuffer *buf = GST_BUFFER_CAST (walk->data);
+
+ GST_DEBUG_OBJECT (dec, "decoding buffer %p, ts %" GST_TIME_FORMAT,
+ buf, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)));
+
+ next = g_list_next (walk);
+ /* decode buffer, resulting data prepended to output queue */
+ gst_buffer_ref (buf);
+ res = gst_base_audio_decoder_chain_forward (dec, buf);
+
+ /* if we generated output, we can discard the buffer, else we
+ * keep it in the queue */
+ if (priv->queued) {
+ GST_DEBUG_OBJECT (dec, "decoded buffer to %p", priv->queued->data);
+ priv->decode = g_list_delete_link (priv->decode, walk);
+ gst_buffer_unref (buf);
+ } else {
+ GST_DEBUG_OBJECT (dec, "buffer did not decode, keeping");
+ }
+ walk = next;
+ }
+
+ /* drain any aggregation (or otherwise) leftover */
+ gst_base_audio_decoder_drain (dec);
+
+ /* now send queued data downstream */
+ while (priv->queued) {
+ GstBuffer *buf = GST_BUFFER_CAST (priv->queued->data);
+
+ if (G_LIKELY (res == GST_FLOW_OK)) {
+ GST_DEBUG_OBJECT (dec, "pushing buffer %p of size %u, "
+ "time %" GST_TIME_FORMAT ", dur %" GST_TIME_FORMAT, buf,
+ GST_BUFFER_SIZE (buf), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
+ GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
+ /* should be already, but let's be sure */
+ buf = gst_buffer_make_metadata_writable (buf);
+ /* avoid stray DISCONT from forward processing,
+ * which have no meaning in reverse pushing */
+ GST_BUFFER_FLAG_UNSET (buf, GST_BUFFER_FLAG_DISCONT);
+ res = gst_pad_push (dec->srcpad, buf);
+ } else {
+ gst_buffer_unref (buf);
+ }
+
+ priv->queued = g_list_delete_link (priv->queued, priv->queued);
+ }
+
+ return res;
+ }
+
+ static GstFlowReturn
+ gst_base_audio_decoder_chain_reverse (GstBaseAudioDecoder * dec,
+ GstBuffer * buf)
+ {
+ GstBaseAudioDecoderPrivate *priv = dec->priv;
+ GstFlowReturn result = GST_FLOW_OK;
+
+ /* if we have a discont, move buffers to the decode list */
+ if (!buf || GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT)) {
+ GST_DEBUG_OBJECT (dec, "received discont");
+ while (priv->gather) {
+ GstBuffer *gbuf;
+
+ gbuf = GST_BUFFER_CAST (priv->gather->data);
+ /* remove from the gather list */
+ priv->gather = g_list_delete_link (priv->gather, priv->gather);
+ /* copy to decode queue */
+ priv->decode = g_list_prepend (priv->decode, gbuf);
+ }
+ /* decode stuff in the decode queue */
+ gst_base_audio_decoder_flush_decode (dec);
+ }
+
+ if (G_LIKELY (buf)) {
+ GST_DEBUG_OBJECT (dec, "gathering buffer %p of size %u, "
+ "time %" GST_TIME_FORMAT ", dur %" GST_TIME_FORMAT, buf,
+ GST_BUFFER_SIZE (buf), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
+ GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
+
+ /* add buffer to gather queue */
+ priv->gather = g_list_prepend (priv->gather, buf);
+ }
+
+ return result;
+ }
+
+ static GstFlowReturn
+ gst_base_audio_decoder_chain (GstPad * pad, GstBuffer * buffer)
+ {
+ GstBaseAudioDecoder *dec;
+ GstFlowReturn ret;
+
+ dec = GST_BASE_AUDIO_DECODER (GST_PAD_PARENT (pad));
+
+ GST_LOG_OBJECT (dec,
+ "received buffer of size %d with ts %" GST_TIME_FORMAT
+ ", duration %" GST_TIME_FORMAT, GST_BUFFER_SIZE (buffer),
+ GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
+ GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)));
+
+ if (GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DISCONT)) {
+ gint64 samples, ts;
+
+ /* track present position */
+ ts = dec->priv->base_ts;
+ samples = dec->priv->samples;
+
+ GST_DEBUG_OBJECT (dec, "handling discont");
+ gst_base_audio_decoder_flush (dec, FALSE);
+ dec->priv->discont = TRUE;
+
+ /* buffer may claim DISCONT loudly, if it can't tell us where we are now,
+ * we'll stick to where we were ...
+ * Particularly useful/needed for upstream BYTE based */
+ if (dec->segment.rate > 0.0 && !GST_BUFFER_TIMESTAMP_IS_VALID (buffer)) {
+ GST_DEBUG_OBJECT (dec, "... but restoring previous ts tracking");
+ dec->priv->base_ts = ts;
+ dec->priv->samples = samples;
+ }
+ }
+
+ if (dec->segment.rate > 0.0)
+ ret = gst_base_audio_decoder_chain_forward (dec, buffer);
+ else
+ ret = gst_base_audio_decoder_chain_reverse (dec, buffer);
+
+ return ret;
+ }
+
+ /* perform upstream byte <-> time conversion (duration, seeking)
+ * if subclass allows and if enough data for moderately decent conversion */
+ static inline gboolean
+ gst_base_audio_decoder_do_byte (GstBaseAudioDecoder * dec)
+ {
+ return dec->priv->ctx.do_byte_time && dec->priv->ctx.info.bpf &&
+ dec->priv->ctx.info.rate <= dec->priv->samples_out;
+ }
+
+ static gboolean
+ gst_base_audio_decoder_sink_eventfunc (GstBaseAudioDecoder * dec,
+ GstEvent * event)
+ {
+ gboolean handled = FALSE;
+
+ switch (GST_EVENT_TYPE (event)) {
+ case GST_EVENT_NEWSEGMENT:
+ {
+ GstFormat format;
+ gdouble rate, arate;
+ gint64 start, stop, time;
+ gboolean update;
+
+ gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
+ &start, &stop, &time);
+
+ if (format == GST_FORMAT_TIME) {
+ GST_DEBUG_OBJECT (dec, "received TIME NEW_SEGMENT %" GST_TIME_FORMAT
+ " -- %" GST_TIME_FORMAT ", time %" GST_TIME_FORMAT
+ ", rate %g, applied_rate %g",
+ GST_TIME_ARGS (start), GST_TIME_ARGS (stop), GST_TIME_ARGS (time),
+ rate, arate);
+ } else {
+ GstFormat dformat = GST_FORMAT_TIME;
+
+ GST_DEBUG_OBJECT (dec, "received NEW_SEGMENT %" G_GINT64_FORMAT
+ " -- %" G_GINT64_FORMAT ", time %" G_GINT64_FORMAT
+ ", rate %g, applied_rate %g", start, stop, time, rate, arate);
+ /* handle newsegment resulting from legacy simple seeking */
+ /* note that we need to convert this whether or not enough data
+ * to handle initial newsegment */
+ if (dec->priv->ctx.do_byte_time &&
+ gst_pad_query_convert (dec->sinkpad, GST_FORMAT_BYTES, start,
+ &dformat, &start)) {
+ /* best attempt convert */
+ /* as these are only estimates, stop is kept open-ended to avoid
+ * premature cutting */
+ GST_DEBUG_OBJECT (dec, "converted to TIME start %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (start));
+ format = GST_FORMAT_TIME;
+ time = start;
+ stop = GST_CLOCK_TIME_NONE;
+ /* replace event */
+ gst_event_unref (event);
+ event = gst_event_new_new_segment_full (update, rate, arate,
+ GST_FORMAT_TIME, start, stop, time);
+ } else {
+ GST_DEBUG_OBJECT (dec, "unsupported format; ignoring");
+ break;
+ }
+ }
+
+ /* finish current segment */
+ gst_base_audio_decoder_drain (dec);
+
+ if (update) {
+ /* time progressed without data, see if we can fill the gap with
+ * some concealment data */
+ GST_DEBUG_OBJECT (dec,
+ "segment update: plc %d, do_plc %d, last_stop %" GST_TIME_FORMAT,
+ dec->priv->plc, dec->priv->ctx.do_plc,
+ GST_TIME_ARGS (dec->segment.last_stop));
+ if (dec->priv->plc && dec->priv->ctx.do_plc &&
+ dec->segment.rate > 0.0 && dec->segment.last_stop < start) {
+ GstBaseAudioDecoderClass *klass;
+ GstBuffer *buf;
+
+ klass = GST_BASE_AUDIO_DECODER_GET_CLASS (dec);
+ /* hand subclass empty frame with duration that needs covering */
+ buf = gst_buffer_new ();
+ GST_BUFFER_DURATION (buf) = start - dec->segment.last_stop;
+ /* best effort, not much error handling */
+ gst_base_audio_decoder_handle_frame (dec, klass, buf);
+ }
+ } else {
+ /* prepare for next one */
+ gst_base_audio_decoder_flush (dec, FALSE);
+ /* and that's where we time from,
+ * in case upstream does not come up with anything better
+ * (e.g. upstream BYTE) */
+ if (format != GST_FORMAT_TIME) {
+ dec->priv->base_ts = start;
+ dec->priv->samples = 0;
+ }
+ }
+
+ /* and follow along with segment */
+ gst_segment_set_newsegment_full (&dec->segment, update, rate, arate,
+ format, start, stop, time);
+
+ gst_pad_push_event (dec->srcpad, event);
+ handled = TRUE;
+ break;
+ }
+
+ case GST_EVENT_FLUSH_START:
+ break;
+
+ case GST_EVENT_FLUSH_STOP:
+ /* prepare for fresh start */
+ gst_base_audio_decoder_flush (dec, TRUE);
+ break;
+
+ case GST_EVENT_EOS:
+ gst_base_audio_decoder_drain (dec);
+ break;
+
+ default:
+ break;
+ }
+
+ return handled;
+ }
+
+ static gboolean
+ gst_base_audio_decoder_sink_event (GstPad * pad, GstEvent * event)
+ {
+ GstBaseAudioDecoder *dec;
+ GstBaseAudioDecoderClass *klass;
+ gboolean handled = FALSE;
+ gboolean ret = TRUE;
+
+ dec = GST_BASE_AUDIO_DECODER (gst_pad_get_parent (pad));
+ klass = GST_BASE_AUDIO_DECODER_GET_CLASS (dec);
+
+ GST_DEBUG_OBJECT (dec, "received event %d, %s", GST_EVENT_TYPE (event),
+ GST_EVENT_TYPE_NAME (event));
+
+ if (klass->event)
+ handled = klass->event (dec, event);
+
+ if (!handled)
+ handled = gst_base_audio_decoder_sink_eventfunc (dec, event);
+
+ if (!handled)
+ ret = gst_pad_event_default (pad, event);
+
+ GST_DEBUG_OBJECT (dec, "event handled");
+
+ gst_object_unref (dec);
+ return ret;
+ }
+
+ static gboolean
+ gst_base_audio_decoder_do_seek (GstBaseAudioDecoder * dec, GstEvent * event)
+ {
+ GstSeekFlags flags;
+ GstSeekType start_type, end_type;
+ GstFormat format;
+ gdouble rate;
+ gint64 start, start_time, end_time;
+ GstSegment seek_segment;
+ guint32 seqnum;
+
+ gst_event_parse_seek (event, &rate, &format, &flags, &start_type,
+ &start_time, &end_type, &end_time);
+
+ /* we'll handle plain open-ended flushing seeks with the simple approach */
+ if (rate != 1.0) {
+ GST_DEBUG_OBJECT (dec, "unsupported seek: rate");
+ return FALSE;
+ }
+
+ if (start_type != GST_SEEK_TYPE_SET) {
+ GST_DEBUG_OBJECT (dec, "unsupported seek: start time");
+ return FALSE;
+ }
+
+ if (end_type != GST_SEEK_TYPE_NONE ||
+ (end_type == GST_SEEK_TYPE_SET && end_time != GST_CLOCK_TIME_NONE)) {
+ GST_DEBUG_OBJECT (dec, "unsupported seek: end time");
+ return FALSE;
+ }
+
+ if (!(flags & GST_SEEK_FLAG_FLUSH)) {
+ GST_DEBUG_OBJECT (dec, "unsupported seek: not flushing");
+ return FALSE;
+ }
+
+ memcpy (&seek_segment, &dec->segment, sizeof (seek_segment));
+ gst_segment_set_seek (&seek_segment, rate, format, flags, start_type,
+ start_time, end_type, end_time, NULL);
+ start_time = seek_segment.last_stop;
+
+ format = GST_FORMAT_BYTES;
+ if (!gst_pad_query_convert (dec->sinkpad, GST_FORMAT_TIME, start_time,
+ &format, &start)) {
+ GST_DEBUG_OBJECT (dec, "conversion failed");
+ return FALSE;
+ }
+
+ seqnum = gst_event_get_seqnum (event);
+ event = gst_event_new_seek (1.0, GST_FORMAT_BYTES, flags,
+ GST_SEEK_TYPE_SET, start, GST_SEEK_TYPE_NONE, -1);
+ gst_event_set_seqnum (event, seqnum);
+
+ GST_DEBUG_OBJECT (dec, "seeking to %" GST_TIME_FORMAT " at byte offset %"
+ G_GINT64_FORMAT, GST_TIME_ARGS (start_time), start);
+
+ return gst_pad_push_event (dec->sinkpad, event);
+ }
+
+ static gboolean
+ gst_base_audio_decoder_src_event (GstPad * pad, GstEvent * event)
+ {
+ GstBaseAudioDecoder *dec;
+ gboolean res = FALSE;
+
+ dec = GST_BASE_AUDIO_DECODER (gst_pad_get_parent (pad));
+
+ GST_DEBUG_OBJECT (dec, "received event %d, %s", GST_EVENT_TYPE (event),
+ GST_EVENT_TYPE_NAME (event));
+
+ switch (GST_EVENT_TYPE (event)) {
+ case GST_EVENT_SEEK:
+ {
+ GstFormat format, tformat;
+ gdouble rate;
+ GstSeekFlags flags;
+ GstSeekType cur_type, stop_type;
+ gint64 cur, stop;
+ gint64 tcur, tstop;
+ guint32 seqnum;
+
+ gst_event_parse_seek (event, &rate, &format, &flags, &cur_type, &cur,
+ &stop_type, &stop);
+ seqnum = gst_event_get_seqnum (event);
+
+ /* upstream gets a chance first */
+ if ((res = gst_pad_push_event (dec->sinkpad, event)))
+ break;
+
+ /* if upstream fails for a time seek, maybe we can help if allowed */
+ if (format == GST_FORMAT_TIME) {
+ if (gst_base_audio_decoder_do_byte (dec))
+ res = gst_base_audio_decoder_do_seek (dec, event);
+ break;
+ }
+
+ /* ... though a non-time seek can be aided as well */
+ /* First bring the requested format to time */
+ tformat = GST_FORMAT_TIME;
+ if (!(res = gst_pad_query_convert (pad, format, cur, &tformat, &tcur)))
+ goto convert_error;
+ if (!(res = gst_pad_query_convert (pad, format, stop, &tformat, &tstop)))
+ goto convert_error;
+
+ /* then seek with time on the peer */
+ event = gst_event_new_seek (rate, GST_FORMAT_TIME,
+ flags, cur_type, tcur, stop_type, tstop);
+ gst_event_set_seqnum (event, seqnum);
+
+ res = gst_pad_push_event (dec->sinkpad, event);
+ break;
+ }
+ default:
+ res = gst_pad_push_event (dec->sinkpad, event);
+ break;
+ }
+ done:
+ gst_object_unref (dec);
+
+ return res;
+
+ /* ERRORS */
+ convert_error:
+ {
+ GST_DEBUG_OBJECT (dec, "cannot convert start/stop for seek");
+ goto done;
+ }
+ }
+
+ /*
+ * gst_base_audio_encoded_audio_convert:
+ * @fmt: audio format of the encoded audio
+ * @bytes: number of encoded bytes
+ * @samples: number of encoded samples
+ * @src_format: source format
+ * @src_value: source value
+ * @dest_format: destination format
+ * @dest_value: destination format
+ *
+ * Helper function to convert @src_value in @src_format to @dest_value in
+ * @dest_format for encoded audio data. Conversion is possible between
+ * BYTE and TIME format by using estimated bitrate based on
+ * @samples and @bytes (and @fmt).
+ */
+ /* FIXME: make gst_base_audio_encoded_audio_convert() public? */
+ static gboolean
+ gst_base_audio_encoded_audio_convert (GstAudioInfo * fmt,
+ gint64 bytes, gint64 samples, GstFormat src_format,
+ gint64 src_value, GstFormat * dest_format, gint64 * dest_value)
+ {
+ gboolean res = FALSE;
+
+ g_return_val_if_fail (dest_format != NULL, FALSE);
+ g_return_val_if_fail (dest_value != NULL, FALSE);
+
+ if (G_UNLIKELY (src_format == *dest_format || src_value == 0 ||
+ src_value == -1)) {
+ if (dest_value)
+ *dest_value = src_value;
+ return TRUE;
+ }
+
+ if (samples == 0 || bytes == 0 || fmt->rate == 0) {
+ GST_DEBUG ("not enough metadata yet to convert");
+ goto exit;
+ }
+
+ bytes *= fmt->rate;
+
+ switch (src_format) {
+ case GST_FORMAT_BYTES:
+ switch (*dest_format) {
+ case GST_FORMAT_TIME:
+ *dest_value = gst_util_uint64_scale (src_value,
+ GST_SECOND * samples, bytes);
+ res = TRUE;
+ break;
+ default:
+ res = FALSE;
+ }
+ break;
+ case GST_FORMAT_TIME:
+ switch (*dest_format) {
+ case GST_FORMAT_BYTES:
+ *dest_value = gst_util_uint64_scale (src_value, bytes,
+ samples * GST_SECOND);
+ res = TRUE;
+ break;
+ default:
+ res = FALSE;
+ }
+ break;
+ default:
+ res = FALSE;
+ }
+
+ exit:
+ return res;
+ }
+
+ static gboolean
+ gst_base_audio_decoder_sink_query (GstPad * pad, GstQuery * query)
+ {
+ gboolean res = TRUE;
+ GstBaseAudioDecoder *dec;
+
+ dec = GST_BASE_AUDIO_DECODER (gst_pad_get_parent (pad));
+
+ switch (GST_QUERY_TYPE (query)) {
+ case GST_QUERY_FORMATS:
+ {
+ gst_query_set_formats (query, 2, GST_FORMAT_TIME, GST_FORMAT_BYTES);
+ res = TRUE;
+ break;
+ }
+ case GST_QUERY_CONVERT:
+ {
+ GstFormat src_fmt, dest_fmt;
+ gint64 src_val, dest_val;
+
+ gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
+ if (!(res = gst_base_audio_encoded_audio_convert (&dec->priv->ctx.info,
+ dec->priv->bytes_in, dec->priv->samples_out,
+ src_fmt, src_val, &dest_fmt, &dest_val)))
+ goto error;
+ gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
+ break;
+ }
+ default:
+ res = gst_pad_query_default (pad, query);
+ break;
+ }
+
+ error:
+ gst_object_unref (dec);
+ return res;
+ }
+
+ static const GstQueryType *
+ gst_base_audio_decoder_get_query_types (GstPad * pad)
+ {
+ static const GstQueryType gst_base_audio_decoder_src_query_types[] = {
+ GST_QUERY_POSITION,
+ GST_QUERY_DURATION,
+ GST_QUERY_CONVERT,
+ GST_QUERY_LATENCY,
+ 0
+ };
+
+ return gst_base_audio_decoder_src_query_types;
+ }
+
+ /* FIXME ? are any of these queries (other than latency) a decoder's business ??
+ * also, the conversion stuff might seem to make sense, but seems to not mind
+ * segment stuff etc at all
+ * Supposedly that's backward compatibility ... */
+ static gboolean
+ gst_base_audio_decoder_src_query (GstPad * pad, GstQuery * query)
+ {
+ GstBaseAudioDecoder *dec;
+ GstPad *peerpad;
+ gboolean res = FALSE;
+
+ dec = GST_BASE_AUDIO_DECODER (GST_PAD_PARENT (pad));
+ peerpad = gst_pad_get_peer (GST_PAD (dec->sinkpad));
+
+ GST_LOG_OBJECT (dec, "handling query: %" GST_PTR_FORMAT, query);
+
+ switch (GST_QUERY_TYPE (query)) {
+ case GST_QUERY_DURATION:
+ {
+ GstFormat format;
+
+ /* upstream in any case */
+ if ((res = gst_pad_query_default (pad, query)))
+ break;
+
+ gst_query_parse_duration (query, &format, NULL);
+ /* try answering TIME by converting from BYTE if subclass allows */
+ if (format == GST_FORMAT_TIME && gst_base_audio_decoder_do_byte (dec)) {
+ gint64 value;
+
+ format = GST_FORMAT_BYTES;
+ if (gst_pad_query_peer_duration (dec->sinkpad, &format, &value)) {
+ GST_LOG_OBJECT (dec, "upstream size %" G_GINT64_FORMAT, value);
+ format = GST_FORMAT_TIME;
+ if (gst_pad_query_convert (dec->sinkpad, GST_FORMAT_BYTES, value,
+ &format, &value)) {
+ gst_query_set_duration (query, GST_FORMAT_TIME, value);
+ res = TRUE;
+ }
+ }
+ }
+ break;
+ }
+ case GST_QUERY_POSITION:
+ {
+ GstFormat format;
+ gint64 time, value;
+
+ if ((res = gst_pad_peer_query (dec->sinkpad, query))) {
+ GST_LOG_OBJECT (dec, "returning peer response");
+ break;
+ }
+
+ /* we start from the last seen time */
+ time = dec->segment.last_stop;
+ /* correct for the segment values */
+ time = gst_segment_to_stream_time (&dec->segment, GST_FORMAT_TIME, time);
+
+ GST_LOG_OBJECT (dec,
+ "query %p: our time: %" GST_TIME_FORMAT, query, GST_TIME_ARGS (time));
+
+ /* and convert to the final format */
+ gst_query_parse_position (query, &format, NULL);
+ if (!(res = gst_pad_query_convert (pad, GST_FORMAT_TIME, time,
+ &format, &value)))
+ break;
+
+ gst_query_set_position (query, format, value);
+
+ GST_LOG_OBJECT (dec,
+ "query %p: we return %" G_GINT64_FORMAT " (format %u)", query, value,
+ format);
+ break;
+ }
+ case GST_QUERY_FORMATS:
+ {
+ gst_query_set_formats (query, 3,
+ GST_FORMAT_TIME, GST_FORMAT_BYTES, GST_FORMAT_DEFAULT);
+ res = TRUE;
+ break;
+ }
+ case GST_QUERY_CONVERT:
+ {
+ GstFormat src_fmt, dest_fmt;
+ gint64 src_val, dest_val;
+
+ gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
+ if (!(res = gst_audio_info_convert (&dec->priv->ctx.info,
+ src_fmt, src_val, dest_fmt, &dest_val)))
+ break;
+ gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
+ break;
+ }
+ case GST_QUERY_LATENCY:
+ {
+ if ((res = gst_pad_peer_query (dec->sinkpad, query))) {
+ gboolean live;
+ GstClockTime min_latency, max_latency;
+
+ gst_query_parse_latency (query, &live, &min_latency, &max_latency);
+ GST_DEBUG_OBJECT (dec, "Peer latency: live %d, min %"
+ GST_TIME_FORMAT " max %" GST_TIME_FORMAT, live,
+ GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
+
+ GST_OBJECT_LOCK (dec);
+ /* add our latency */
+ if (min_latency != -1)
+ min_latency += dec->priv->ctx.min_latency;
+ if (max_latency != -1)
+ max_latency += dec->priv->ctx.max_latency;
+ GST_OBJECT_UNLOCK (dec);
+
+ gst_query_set_latency (query, live, min_latency, max_latency);
+ }
+ break;
+ }
+ default:
+ res = gst_pad_query_default (pad, query);
+ break;
+ }
+
+ gst_object_unref (peerpad);
+ return res;
+ }
+
+ static gboolean
+ gst_base_audio_decoder_stop (GstBaseAudioDecoder * dec)
+ {
+ GstBaseAudioDecoderClass *klass;
+ gboolean ret = TRUE;
+
+ GST_DEBUG_OBJECT (dec, "gst_base_audio_decoder_stop");
+
+ klass = GST_BASE_AUDIO_DECODER_GET_CLASS (dec);
+
+ if (klass->stop) {
+ ret = klass->stop (dec);
+ }
+
+ /* clean up */
+ gst_base_audio_decoder_reset (dec, TRUE);
+
+ if (ret)
+ dec->priv->active = FALSE;
+
+ return TRUE;
+ }
+
+ static gboolean
+ gst_base_audio_decoder_start (GstBaseAudioDecoder * dec)
+ {
+ GstBaseAudioDecoderClass *klass;
+ gboolean ret = TRUE;
+
+ GST_DEBUG_OBJECT (dec, "gst_base_audio_decoder_start");
+
+ klass = GST_BASE_AUDIO_DECODER_GET_CLASS (dec);
+
+ /* arrange clean state */
+ gst_base_audio_decoder_reset (dec, TRUE);
+
+ if (klass->start) {
+ ret = klass->start (dec);
+ }
+
+ if (ret)
+ dec->priv->active = TRUE;
+
+ return TRUE;
+ }
+
+ static void
+ gst_base_audio_decoder_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec)
+ {
+ GstBaseAudioDecoder *dec;
+
+ dec = GST_BASE_AUDIO_DECODER (object);
+
+ switch (prop_id) {
+ case PROP_LATENCY:
+ g_value_set_int64 (value, dec->priv->latency);
+ break;
+ case PROP_TOLERANCE:
+ g_value_set_int64 (value, dec->priv->tolerance);
+ break;
+ case PROP_PLC:
+ g_value_set_boolean (value, dec->priv->plc);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+ }
+
+ static void
+ gst_base_audio_decoder_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec)
+ {
+ GstBaseAudioDecoder *dec;
+
+ dec = GST_BASE_AUDIO_DECODER (object);
+
+ switch (prop_id) {
+ case PROP_LATENCY:
+ dec->priv->latency = g_value_get_int64 (value);
+ break;
+ case PROP_TOLERANCE:
+ dec->priv->tolerance = g_value_get_int64 (value);
+ break;
+ case PROP_PLC:
+ dec->priv->plc = g_value_get_boolean (value);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+ }
+
+ static GstStateChangeReturn
+ gst_base_audio_decoder_change_state (GstElement * element,
+ GstStateChange transition)
+ {
+ GstBaseAudioDecoder *codec;
+ GstStateChangeReturn ret;
+
+ codec = GST_BASE_AUDIO_DECODER (element);
+
+ switch (transition) {
+ case GST_STATE_CHANGE_NULL_TO_READY:
+ break;
+ case GST_STATE_CHANGE_READY_TO_PAUSED:
+ if (!gst_base_audio_decoder_start (codec)) {
+ goto start_failed;
+ }
+ break;
+ case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
+ break;
+ default:
+ break;
+ }
+
+ ret = parent_class->change_state (element, transition);
+
+ switch (transition) {
+ case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
+ break;
+ case GST_STATE_CHANGE_PAUSED_TO_READY:
+ if (!gst_base_audio_decoder_stop (codec)) {
+ goto stop_failed;
+ }
+ break;
+ case GST_STATE_CHANGE_READY_TO_NULL:
+ break;
+ default:
+ break;
+ }
+
+ return ret;
+
+ start_failed:
+ {
+ GST_ELEMENT_ERROR (codec, LIBRARY, INIT, (NULL), ("Failed to start codec"));
+ return GST_STATE_CHANGE_FAILURE;
+ }
+ stop_failed:
+ {
+ GST_ELEMENT_ERROR (codec, LIBRARY, INIT, (NULL), ("Failed to stop codec"));
+ return GST_STATE_CHANGE_FAILURE;
+ }
+ }
+
+ GstFlowReturn
+ _gst_base_audio_decoder_error (GstBaseAudioDecoder * dec, gint weight,
+ GQuark domain, gint code, gchar * txt, gchar * dbg, const gchar * file,
+ const gchar * function, gint line)
+ {
+ if (txt)
+ GST_WARNING_OBJECT (dec, "error: %s", txt);
+ if (dbg)
+ GST_WARNING_OBJECT (dec, "error: %s", dbg);
+ dec->priv->error_count += weight;
+ dec->priv->discont = TRUE;
+ if (dec->priv->ctx.max_errors < dec->priv->error_count) {
+ gst_element_message_full (GST_ELEMENT (dec), GST_MESSAGE_ERROR,
+ domain, code, txt, dbg, file, function, line);
+ return GST_FLOW_ERROR;
+ } else {
+ return GST_FLOW_OK;
+ }
+ }
+
+ /**
+ * gst_base_audio_decoder_get_audio_info:
+ * @dec: a #GstBaseAudioDecoder
+ *
+ * Returns: a #GstAudioInfo describing the input audio format
+ *
+ * Since: 0.10.36
+ */
+ GstAudioInfo *
+ gst_base_audio_decoder_get_audio_info (GstBaseAudioDecoder * dec)
+ {
+ g_return_val_if_fail (GST_IS_BASE_AUDIO_DECODER (dec), NULL);
+
+ return &dec->priv->ctx.info;
+ }
+
+ /**
+ * gst_base_audio_decoder_set_plc_aware:
+ * @dec: a #GstBaseAudioDecoder
+ * @plc: new plc state
+ *
+ * Indicates whether or not subclass handles packet loss concealment (plc).
+ *
+ * Since: 0.10.36
+ */
+ void
+ gst_base_audio_decoder_set_plc_aware (GstBaseAudioDecoder * dec, gboolean plc)
+ {
+ g_return_if_fail (GST_IS_BASE_AUDIO_DECODER (dec));
+
+ dec->priv->ctx.do_plc = plc;
+ }
+
+ /**
+ * gst_base_audio_decoder_get_plc_aware:
+ * @dec: a #GstBaseAudioDecoder
+ *
+ * Returns: currently configured plc handling
+ *
+ * Since: 0.10.36
+ */
+ gint
+ gst_base_audio_decoder_get_plc_aware (GstBaseAudioDecoder * dec)
+ {
+ g_return_val_if_fail (GST_IS_BASE_AUDIO_DECODER (dec), 0);
+
+ return dec->priv->ctx.do_plc;
+ }
+
+ /**
+ * gst_base_audio_decoder_set_byte_time:
+ * @dec: a #GstBaseAudioDecoder
+ * @enabled: whether to enable byte to time conversion
+ *
+ * Allows baseclass to perform byte to time estimated conversion.
+ *
+ * Since: 0.10.36
+ */
+ void
+ gst_base_audio_decoder_set_byte_time (GstBaseAudioDecoder * dec,
+ gboolean enabled)
+ {
+ g_return_if_fail (GST_IS_BASE_AUDIO_DECODER (dec));
+
+ dec->priv->ctx.do_byte_time = enabled;
+ }
+
+ /**
+ * gst_base_audio_decoder_get_byte_time:
+ * @dec: a #GstBaseAudioDecoder
+ *
+ * Returns: currently configured byte to time conversion setting
+ *
+ * Since: 0.10.36
+ */
+ gint
+ gst_base_audio_decoder_get_byte_time (GstBaseAudioDecoder * dec)
+ {
+ g_return_val_if_fail (GST_IS_BASE_AUDIO_DECODER (dec), 0);
+
+ return dec->priv->ctx.do_byte_time;
+ }
+
+ /**
+ * gst_base_audio_decoder_get_delay:
+ * @dec: a #GstBaseAudioDecoder
+ *
+ * Returns: currently configured decoder delay
+ *
+ * Since: 0.10.36
+ */
+ gint
+ gst_base_audio_decoder_get_delay (GstBaseAudioDecoder * dec)
+ {
+ g_return_val_if_fail (GST_IS_BASE_AUDIO_DECODER (dec), 0);
+
+ return dec->priv->ctx.delay;
+ }
+
+ /**
+ * gst_base_audio_decoder_set_max_errors:
+ * @dec: a #GstBaseAudioDecoder
+ * @num: max tolerated errors
+ *
+ * Sets numbers of tolerated decoder errors, where a tolerated one is then only
+ * warned about, but more than tolerated will lead to fatal error.
+ *
+ * Since: 0.10.36
+ */
+ void
+ gst_base_audio_decoder_set_max_errors (GstBaseAudioDecoder * enc, gint num)
+ {
+ g_return_if_fail (GST_IS_BASE_AUDIO_DECODER (enc));
+
+ enc->priv->ctx.max_errors = num;
+ }
+
+ /**
+ * gst_base_audio_decoder_get_max_errors:
+ * @dec: a #GstBaseAudioDecoder
+ *
+ * Returns: currently configured decoder tolerated error count.
+ *
+ * Since: 0.10.36
+ */
+ gint
+ gst_base_audio_decoder_get_max_errors (GstBaseAudioDecoder * dec)
+ {
+ g_return_val_if_fail (GST_IS_BASE_AUDIO_DECODER (dec), 0);
+
+ return dec->priv->ctx.max_errors;
+ }
+
+ /**
+ * gst_base_audio_decoder_set_latency:
+ * @dec: a #GstBaseAudioDecoder
+ * @min: minimum latency
+ * @max: maximum latency
+ *
+ * Sets decoder latency.
+ *
+ * Since: 0.10.36
+ */
+ void
+ gst_base_audio_decoder_set_latency (GstBaseAudioDecoder * dec,
+ GstClockTime min, GstClockTime max)
+ {
+ g_return_if_fail (GST_IS_BASE_AUDIO_DECODER (dec));
+
+ GST_OBJECT_LOCK (dec);
+ dec->priv->ctx.min_latency = min;
+ dec->priv->ctx.max_latency = max;
+ GST_OBJECT_UNLOCK (dec);
+ }
+
+ /**
+ * gst_base_audio_decoder_get_latency:
+ * @dec: a #GstBaseAudioDecoder
+ * @min: a pointer to storage to hold minimum latency
+ * @max: a pointer to storage to hold maximum latency
+ *
+ * Returns currently configured latency.
+ *
+ * Since: 0.10.36
+ */
+ void
+ gst_base_audio_decoder_get_latency (GstBaseAudioDecoder * dec,
+ GstClockTime * min, GstClockTime * max)
+ {
+ g_return_if_fail (GST_IS_BASE_AUDIO_DECODER (dec));
+
+ GST_OBJECT_LOCK (dec);
+ if (min)
+ *min = dec->priv->ctx.min_latency;
+ if (max)
+ *max = dec->priv->ctx.max_latency;
+ GST_OBJECT_UNLOCK (dec);
+ }
+
+ /**
+ * gst_base_audio_decoder_get_parse_state:
+ * @dec: a #GstBaseAudioDecoder
+ * @min: a pointer to storage to hold current sync state
+ * @max: a pointer to storage to hold current eos state
+ *
+ * Return current parsing (sync and eos) state.
+ *
+ * Since: 0.10.36
+ */
+ void
+ gst_base_audio_decoder_get_parse_state (GstBaseAudioDecoder * dec,
+ gboolean * sync, gboolean * eos)
+ {
+ g_return_if_fail (GST_IS_BASE_AUDIO_DECODER (dec));
+
+ if (sync)
+ *sync = dec->priv->ctx.sync;
+ if (eos)
+ *eos = dec->priv->ctx.eos;
+ }
+
+ /**
+ * gst_base_audio_decoder_set_plc:
+ * @dec: a #GstBaseAudioDecoder
+ * @enabled: new state
+ *
+ * Enable or disable decoder packet loss concealment, provided subclass
+ * and codec are capable and allow handling plc.
+ *
+ * MT safe.
+ *
+ * Since: 0.10.36
+ */
+ void
+ gst_base_audio_decoder_set_plc (GstBaseAudioDecoder * dec, gboolean enabled)
+ {
+ g_return_if_fail (GST_IS_BASE_AUDIO_DECODER (dec));
+
+ GST_LOG_OBJECT (dec, "enabled: %d", enabled);
+
+ GST_OBJECT_LOCK (dec);
+ dec->priv->plc = enabled;
+ GST_OBJECT_UNLOCK (dec);
+ }
+
+ /**
+ * gst_base_audio_decoder_get_plc:
+ * @dec: a #GstBaseAudioDecoder
+ *
+ * Queries decoder packet loss concealment handling.
+ *
+ * Returns: TRUE if packet loss concealment is enabled.
+ *
+ * MT safe.
+ *
+ * Since: 0.10.36
+ */
+ gboolean
+ gst_base_audio_decoder_get_plc (GstBaseAudioDecoder * dec)
+ {
+ gboolean result;
+
+ g_return_val_if_fail (GST_IS_BASE_AUDIO_DECODER (dec), FALSE);
+
+ GST_OBJECT_LOCK (dec);
+ result = dec->priv->plc;
+ GST_OBJECT_UNLOCK (dec);
+
+ return result;
+ }
+
+ /**
+ * gst_base_audio_decoder_set_min_latency:
+ * @dec: a #GstBaseAudioDecoder
+ * @num: new minimum latency
+ *
+ * Sets decoder minimum aggregation latency.
+ *
+ * MT safe.
+ *
+ * Since: 0.10.36
+ */
+ void
+ gst_base_audio_decoder_set_min_latency (GstBaseAudioDecoder * dec, gint64 num)
+ {
+ g_return_if_fail (GST_IS_BASE_AUDIO_DECODER (dec));
+
+ GST_OBJECT_LOCK (dec);
+ dec->priv->latency = num;
+ GST_OBJECT_UNLOCK (dec);
+ }
+
+ /**
+ * gst_base_audio_decoder_get_min_latency:
+ * @enc: a #GstBaseAudioDecoder
+ *
+ * Queries decoder's latency aggregation.
+ *
+ * Returns: aggregation latency.
+ *
+ * MT safe.
+ *
+ * Since: 0.10.36
+ */
+ gint64
+ gst_base_audio_decoder_get_min_latency (GstBaseAudioDecoder * dec)
+ {
+ gint64 result;
+
+ g_return_val_if_fail (GST_IS_BASE_AUDIO_DECODER (dec), FALSE);
+
+ GST_OBJECT_LOCK (dec);
+ result = dec->priv->latency;
+ GST_OBJECT_UNLOCK (dec);
+
+ return result;
+ }
+
+ /**
+ * gst_base_audio_decoder_set_tolerance:
+ * @dec: a #GstBaseAudioDecoder
+ * @tolerance: new tolerance
+ *
+ * Configures decoder audio jitter tolerance threshold.
+ *
+ * MT safe.
+ *
+ * Since: 0.10.36
+ */
+ void
+ gst_base_audio_decoder_set_tolerance (GstBaseAudioDecoder * dec,
+ gint64 tolerance)
+ {
+ g_return_if_fail (GST_IS_BASE_AUDIO_DECODER (dec));
+
+ GST_OBJECT_LOCK (dec);
+ dec->priv->tolerance = tolerance;
+ GST_OBJECT_UNLOCK (dec);
+ }
+
+ /**
+ * gst_base_audio_decoder_get_tolerance:
+ * @dec: a #GstBaseAudioDecoder
+ *
+ * Queries current audio jitter tolerance threshold.
+ *
+ * Returns: decoder audio jitter tolerance threshold.
+ *
+ * MT safe.
+ *
+ * Since: 0.10.36
+ */
+ gint64
+ gst_base_audio_decoder_get_tolerance (GstBaseAudioDecoder * dec)
+ {
+ gint64 result;
+
+ g_return_val_if_fail (GST_IS_BASE_AUDIO_DECODER (dec), 0);
+
+ GST_OBJECT_LOCK (dec);
+ result = dec->priv->tolerance;
+ GST_OBJECT_UNLOCK (dec);
+
+ return result;
+ }
projects / platform / upstream / gstreamer.git / commitdiff