From: Wim Taymans Date: Mon, 29 Aug 2011 09:37:36 +0000 (+0200) Subject: Merge branch 'master' into 0.11 X-Git-Tag: 1.19.3~511^2~7315 X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=e1287b97abb8f9edfcf43db72d9dc440cafa16e3;p=platform%2Fupstream%2Fgstreamer.git Merge branch 'master' into 0.11 Conflicts: ext/ogg/gstoggmux.c gst-libs/gst/audio/audio.c gst-libs/gst/audio/audio.h gst-libs/gst/audio/multichannel.h gst-libs/gst/pbutils/Makefile.am gst-libs/gst/pbutils/gstdiscoverer.c gst/playback/gstplaysinkaudioconvert.c gst/playback/gstplaysinkvideoconvert.c win32/common/libgstaudio.def --- e1287b97abb8f9edfcf43db72d9dc440cafa16e3 diff --cc gst-libs/gst/audio/audio.c index 2416c58,23fe19e..fda6e07 --- a/gst-libs/gst/audio/audio.c +++ b/gst-libs/gst/audio/audio.c @@@ -359,151 -218,489 +359,582 @@@ no_channels } /** - * 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. diff --cc gst-libs/gst/audio/gstbaseaudiodecoder.c index 0000000,08441b9..637e140 mode 000000,100644..100644 --- a/gst-libs/gst/audio/gstbaseaudiodecoder.c +++ b/gst-libs/gst/audio/gstbaseaudiodecoder.c @@@ -1,0 -1,2320 +1,2320 @@@ + /* GStreamer + * Copyright (C) 2009 Igalia S.L. + * Author: Iago Toral Quiroga + * Copyright (C) 2011 Mark Nauwelaerts . + * Copyright (C) 2011 Nokia Corporation. All rights reserved. + * Contact: Stefan Kost + * + * 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. + * + * + * Configuration + * + * 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). + * + * + * 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. + * + * + * GstBaseAudioDecoder calls @stop at end of all processing. + * + * + * + * 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. + * + * + * Data processing + * + * 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. + * + * + * Input frame is provided to subclass' @handle_frame. + * + * + * If codec processing results in decoded data, subclass should call + * @gst_base_audio_decoder_finish_frame to have decoded data pushed + * downstream. + * + * + * 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. + * + * + * 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. + * + * + * + * + * Shutdown phase + * + * GstBaseAudioDecoder class calls @stop to inform the subclass that data + * parsing will be stopped. + * + * + * + * + * + * 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: + * + * Provide pad templates + * + * Set source pad caps when appropriate + * + * + * Set user-configurable properties to sane defaults for format and + * implementing codec at hand, and convey some subclass capabilities and + * expectations in context. + * + * + * 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. + * + * + */ + + #ifdef HAVE_CONFIG_H + #include "config.h" + #endif + + #define GST_USE_UNSTABLE_API + #include "gstbaseaudiodecoder.h" + #include + + #include + + 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; ++ 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; + } diff --cc gst-libs/gst/pbutils/Makefile.am index abb4fcd,1ed49ab..16ff053 --- a/gst-libs/gst/pbutils/Makefile.am +++ b/gst-libs/gst/pbutils/Makefile.am @@@ -82,10 -79,8 +82,9 @@@ GstPbutils-@GST_MAJORMINOR@.gir: $(INTR --nsversion=@GST_MAJORMINOR@ \ --strip-prefix=Gst \ $(gir_cincludes) \ + -DGST_USE_UNSTABLE_API \ -I$(top_srcdir)/gst-libs \ -I$(top_builddir)/gst-libs \ - --add-include-path=$(srcdir)/../video \ --add-include-path=`$(PKG_CONFIG) --variable=girdir gstreamer-@GST_MAJORMINOR@` \ --library=libgstpbutils-@GST_MAJORMINOR@.la \ --library-path=`$(PKG_CONFIG) --variable=libdir gstreamer-@GST_MAJORMINOR@` \ diff --cc gst-libs/gst/pbutils/gstdiscoverer.c index bdbcb26,bd5ad95..05bd86e --- a/gst-libs/gst/pbutils/gstdiscoverer.c +++ b/gst-libs/gst/pbutils/gstdiscoverer.c @@@ -435,16 -433,12 +434,12 @@@ _event_probe (GstPad * pad, GstProbeTyp DISCO_UNLOCK (ps->dc); } - return TRUE; + return GST_PROBE_OK; } - static void - uridecodebin_pad_added_cb (GstElement * uridecodebin, GstPad * pad, - GstDiscoverer * dc) + static gboolean + is_subtitle_caps (const GstCaps * caps) { - PrivateStream *ps; - GstPad *sinkpad = NULL; - GstCaps *caps; static GstCaps *subs_caps = NULL; if (!subs_caps) { @@@ -469,10 -474,10 +475,10 @@@ uridecodebin_pad_added_cb (GstElement g_object_set (ps->sink, "silent", TRUE, NULL); g_object_set (ps->queue, "max-size-buffers", 1, "silent", TRUE, NULL); - caps = gst_pad_get_caps_reffed (pad); + caps = gst_pad_get_caps (pad, NULL); - if (gst_caps_can_intersect (caps, subs_caps)) { - /* Subtitle streams are sparse and don't provide any information - don't + if (is_subtitle_caps (caps)) { + /* Subtitle streams are sparse and may not provide any information - don't * wait for data to preroll */ g_object_set (ps->sink, "async", FALSE, NULL); } diff --cc gst-libs/gst/pbutils/pbutils-private.h index 2d7fcd8,2efaa17..8ac6147 --- a/gst-libs/gst/pbutils/pbutils-private.h +++ b/gst-libs/gst/pbutils/pbutils-private.h @@@ -64,8 -66,14 +66,14 @@@ struct _GstDiscovererVideoInfo gboolean is_image; }; + struct _GstDiscovererSubtitleInfo { + GstDiscovererStreamInfo parent; + + gchar *language; + }; + struct _GstDiscovererInfo { - GstMiniObject parent; + GObject parent; gchar *uri; GstDiscovererResult result; diff --cc gst-libs/gst/rtp/gstbasertppayload.c index 112f9a4,d7f1aa3..3bca852 --- a/gst-libs/gst/rtp/gstbasertppayload.c +++ b/gst-libs/gst/rtp/gstbasertppayload.c @@@ -391,40 -409,33 +396,42 @@@ gst_basertppayload_event_default (GstBa GstSegment *segment; segment = &basertppayload->segment; - - gst_event_parse_new_segment_full (event, &update, &rate, &arate, &fmt, - &start, &stop, &position); - gst_segment_set_newsegment_full (segment, update, rate, arate, fmt, start, - stop, position); + gst_event_copy_segment (event, segment); + basertppayload->priv->base_offset = GST_BUFFER_OFFSET_NONE; + GST_DEBUG_OBJECT (basertppayload, - "configured NEWSEGMENT update %d, rate %lf, applied rate %lf, " - "format %d, " - "%" G_GINT64_FORMAT " -- %" G_GINT64_FORMAT ", time %" - G_GINT64_FORMAT ", accum %" G_GINT64_FORMAT, update, rate, arate, - segment->format, segment->start, segment->stop, segment->time, - segment->accum); - /* fallthrough */ + "configured SEGMENT %" GST_SEGMENT_FORMAT, segment); + res = gst_pad_event_default (basertppayload->sinkpad, event); + break; } default: - res = gst_pad_event_default (pad, event); + res = gst_pad_event_default (basertppayload->sinkpad, event); break; } + return res; +} + +static gboolean +gst_basertppayload_event (GstPad * pad, GstEvent * event) +{ + GstBaseRTPPayload *basertppayload; + GstBaseRTPPayloadClass *basertppayload_class; + gboolean res = FALSE; + + basertppayload = GST_BASE_RTP_PAYLOAD (gst_pad_get_parent (pad)); + if (G_UNLIKELY (basertppayload == NULL)) { + gst_event_unref (event); + return FALSE; + } + + basertppayload_class = GST_BASE_RTP_PAYLOAD_GET_CLASS (basertppayload); + + if (basertppayload_class->handle_event) + res = basertppayload_class->handle_event (basertppayload, event); + else + gst_event_unref (event); -done: gst_object_unref (basertppayload); return res; diff --cc gst/playback/gstplaysink.c index e0a540b,01f00a5..f95e3e9 --- a/gst/playback/gstplaysink.c +++ b/gst/playback/gstplaysink.c @@@ -3020,14 -2984,14 +3024,14 @@@ caps_notify_cb (GstPad * pad, GParamSpe if (pad == playsink->audio_pad) { raw = is_raw_pad (pad); -- reconfigure = (! !playsink->audio_pad_raw != ! !raw) ++ reconfigure = (!!playsink->audio_pad_raw != !!raw) && playsink->audiochain; GST_DEBUG_OBJECT (pad, "Audio caps changed: raw %d reconfigure %d caps %" GST_PTR_FORMAT, raw, reconfigure, caps); } else if (pad == playsink->video_pad) { raw = is_raw_pad (pad); -- reconfigure = (! !playsink->video_pad_raw != ! !raw) ++ reconfigure = (!!playsink->video_pad_raw != !!raw) && playsink->videochain; GST_DEBUG_OBJECT (pad, "Video caps changed: raw %d reconfigure %d caps %" GST_PTR_FORMAT, raw, diff --cc gst/playback/gstplaysinkaudioconvert.c index 08f356e,8e405d2..c409d02 --- a/gst/playback/gstplaysinkaudioconvert.c +++ b/gst/playback/gstplaysinkaudioconvert.c @@@ -72,11 -72,31 +72,33 @@@ post_missing_element_message (GstPlaySi gst_element_post_message (GST_ELEMENT_CAST (self), msg); } + static void + distribute_running_time (GstElement * element, const GstSegment * segment) + { + GstEvent *event; + GstPad *pad; + + pad = gst_element_get_static_pad (element, "sink"); + + if (segment->accum) { + event = gst_event_new_new_segment_full (FALSE, segment->rate, + segment->applied_rate, segment->format, 0, segment->accum, 0); + gst_pad_send_event (pad, event); + } + + event = gst_event_new_new_segment_full (FALSE, segment->rate, + segment->applied_rate, segment->format, + segment->start, segment->stop, segment->time); + gst_pad_send_event (pad, event); + + gst_object_unref (pad); + } + -static void -pad_blocked_cb (GstPad * pad, gboolean blocked, GstPlaySinkAudioConvert * self) +static GstProbeReturn +pad_blocked_cb (GstPad * pad, GstProbeType type, gpointer type_data, + gpointer user_data) { + GstPlaySinkAudioConvert *self = user_data; GstPad *peer; GstCaps *caps; gboolean raw; diff --cc gst/playback/gstplaysinkvideoconvert.c index 494d44a,19986fe..a8c710d --- a/gst/playback/gstplaysinkvideoconvert.c +++ b/gst/playback/gstplaysinkvideoconvert.c @@@ -72,11 -72,31 +72,33 @@@ post_missing_element_message (GstPlaySi gst_element_post_message (GST_ELEMENT_CAST (self), msg); } + static void + distribute_running_time (GstElement * element, const GstSegment * segment) + { + GstEvent *event; + GstPad *pad; + + pad = gst_element_get_static_pad (element, "sink"); + + if (segment->accum) { + event = gst_event_new_new_segment_full (FALSE, segment->rate, + segment->applied_rate, segment->format, 0, segment->accum, 0); + gst_pad_send_event (pad, event); + } + + event = gst_event_new_new_segment_full (FALSE, segment->rate, + segment->applied_rate, segment->format, + segment->start, segment->stop, segment->time); + gst_pad_send_event (pad, event); + + gst_object_unref (pad); + } + -static void -pad_blocked_cb (GstPad * pad, gboolean blocked, GstPlaySinkVideoConvert * self) +static GstProbeReturn +pad_blocked_cb (GstPad * pad, GstProbeType type, gpointer type_data, + gpointer user_data) { + GstPlaySinkVideoConvert *self = user_data; GstPad *peer; GstCaps *caps; gboolean raw; diff --cc win32/common/libgstaudio.def index 1e2fff4,414a967..568f033 --- a/win32/common/libgstaudio.def +++ b/win32/common/libgstaudio.def @@@ -31,6 -33,46 +32,47 @@@ EXPORT gst_audio_set_structure_channel_positions_list gst_audio_sink_get_type gst_audio_src_get_type + gst_audio_structure_set_int + gst_base_audio_decoder_finish_frame + gst_base_audio_decoder_get_audio_info + gst_base_audio_decoder_get_byte_time + gst_base_audio_decoder_get_delay + gst_base_audio_decoder_get_latency + gst_base_audio_decoder_get_max_errors + gst_base_audio_decoder_get_min_latency + gst_base_audio_decoder_get_parse_state + gst_base_audio_decoder_get_plc + gst_base_audio_decoder_get_plc_aware + gst_base_audio_decoder_get_tolerance + gst_base_audio_decoder_get_type + gst_base_audio_decoder_set_byte_time + gst_base_audio_decoder_set_latency + gst_base_audio_decoder_set_max_errors + gst_base_audio_decoder_set_min_latency + gst_base_audio_decoder_set_plc + gst_base_audio_decoder_set_plc_aware + gst_base_audio_decoder_set_tolerance + gst_base_audio_encoder_finish_frame + gst_base_audio_encoder_get_audio_info + gst_base_audio_encoder_get_frame_max + gst_base_audio_encoder_get_frame_samples + gst_base_audio_encoder_get_hard_resync + gst_base_audio_encoder_get_latency + gst_base_audio_encoder_get_lookahead + gst_base_audio_encoder_get_mark_granule + gst_base_audio_encoder_get_perfect_timestamp + gst_base_audio_encoder_get_tolerance + gst_base_audio_encoder_get_type + gst_base_audio_encoder_proxy_getcaps + gst_base_audio_encoder_set_frame_max + gst_base_audio_encoder_set_frame_samples + gst_base_audio_encoder_set_hard_resync + gst_base_audio_encoder_set_latency + gst_base_audio_encoder_set_lookahead + gst_base_audio_encoder_set_mark_granule + gst_base_audio_encoder_set_perfect_timestamp + gst_base_audio_encoder_set_tolerance ++>>>>>>> master gst_base_audio_sink_create_ringbuffer gst_base_audio_sink_get_drift_tolerance gst_base_audio_sink_get_provide_clock