* qatar/master: libopus: Remap channels using libopus' internal remapping. Opus decoder using libopus avcodec: document the use of AVCodecContext.delay for audio decoding vc1dec: add flush function for WMV9 and VC-1 decoders http: Increase buffer sizes to cope with longer URIs nutenc: const correctness for ff_put_v_trace/put_s_trace function arguments h264_refs: Fix debug tprintf argument types golomb: const correctness for get_ue()/get_se() function arguments get_bits: const correctness for get_bits_trace()/get_xbits_trace() arguments Conflicts: Changelog libavcodec/Makefile libavcodec/version.h libavformat/http.c Merged-by: Michael Niedermayer <michaelni@gmx.at>tags/n1.1
| @@ -813,6 +813,8 @@ following image formats are supported: | |||
| @item Musepack SV7 @tab @tab X | |||
| @item Musepack SV8 @tab @tab X | |||
| @item Nellymoser Asao @tab X @tab X | |||
| @item Opus @tab @tab E | |||
| @tab supported through external library libopus | |||
| @item PCM A-law @tab X @tab X | |||
| @item PCM mu-law @tab X @tab X | |||
| @item PCM 16-bit little-endian planar @tab @tab X | |||
| @@ -657,7 +657,7 @@ OBJS-$(CONFIG_LIBOPENCORE_AMRNB_ENCODER) += libopencore-amr.o \ | |||
| OBJS-$(CONFIG_LIBOPENCORE_AMRWB_DECODER) += libopencore-amr.o | |||
| OBJS-$(CONFIG_LIBOPENJPEG_DECODER) += libopenjpegdec.o | |||
| OBJS-$(CONFIG_LIBOPENJPEG_ENCODER) += libopenjpegenc.o | |||
| OBJS-$(CONFIG_LIBOPUS_DECODER) += libopus_dec.o vorbis_data.o | |||
| OBJS-$(CONFIG_LIBOPUS_DECODER) += libopusdec.o vorbis_data.o | |||
| OBJS-$(CONFIG_LIBSCHROEDINGER_DECODER) += libschroedingerdec.o \ | |||
| libschroedinger.o | |||
| OBJS-$(CONFIG_LIBSCHROEDINGER_ENCODER) += libschroedingerenc.o \ | |||
| @@ -1608,12 +1608,15 @@ typedef struct AVCodecContext { | |||
| * encoded input. | |||
| * | |||
| * Audio: | |||
| * Number of "priming" samples added to the beginning of the stream | |||
| * during encoding. The decoded output will be delayed by this many | |||
| * samples relative to the input to the encoder. Note that this field is | |||
| * purely informational and does not directly affect the pts output by | |||
| * the encoder, which should always be based on the actual presentation | |||
| * time, including any delay. | |||
| * For encoding, this is the number of "priming" samples added to the | |||
| * beginning of the stream. The decoded output will be delayed by this | |||
| * many samples relative to the input to the encoder. Note that this | |||
| * field is purely informational and does not directly affect the pts | |||
| * output by the encoder, which should always be based on the actual | |||
| * presentation time, including any delay. | |||
| * For decoding, this is the number of samples the decoder needs to | |||
| * output before the decoder's output is valid. When seeking, you should | |||
| * start decoding this many samples prior to your desired seek point. | |||
| * | |||
| * - encoding: Set by libavcodec. | |||
| * - decoding: Set by libavcodec. | |||
| @@ -521,7 +521,7 @@ static inline void print_bin(int bits, int n) | |||
| av_log(NULL, AV_LOG_DEBUG, " "); | |||
| } | |||
| static inline int get_bits_trace(GetBitContext *s, int n, char *file, | |||
| static inline int get_bits_trace(GetBitContext *s, int n, const char *file, | |||
| const char *func, int line) | |||
| { | |||
| int r = get_bits(s, n); | |||
| @@ -532,7 +532,7 @@ static inline int get_bits_trace(GetBitContext *s, int n, char *file, | |||
| return r; | |||
| } | |||
| static inline int get_vlc_trace(GetBitContext *s, VLC_TYPE (*table)[2], | |||
| int bits, int max_depth, char *file, | |||
| int bits, int max_depth, const char *file, | |||
| const char *func, int line) | |||
| { | |||
| int show = show_bits(s, 24); | |||
| @@ -547,7 +547,7 @@ static inline int get_vlc_trace(GetBitContext *s, VLC_TYPE (*table)[2], | |||
| bits2, len, r, pos, file, func, line); | |||
| return r; | |||
| } | |||
| static inline int get_xbits_trace(GetBitContext *s, int n, char *file, | |||
| static inline int get_xbits_trace(GetBitContext *s, int n, const char *file, | |||
| const char *func, int line) | |||
| { | |||
| int show = show_bits(s, n); | |||
| @@ -374,7 +374,9 @@ static inline int get_sr_golomb_shorten(GetBitContext* gb, int k) | |||
| #ifdef TRACE | |||
| static inline int get_ue(GetBitContext *s, char *file, const char *func, int line){ | |||
| static inline int get_ue(GetBitContext *s, const char *file, const char *func, | |||
| int line) | |||
| { | |||
| int show= show_bits(s, 24); | |||
| int pos= get_bits_count(s); | |||
| int i= get_ue_golomb(s); | |||
| @@ -388,7 +390,9 @@ static inline int get_ue(GetBitContext *s, char *file, const char *func, int lin | |||
| return i; | |||
| } | |||
| static inline int get_se(GetBitContext *s, char *file, const char *func, int line){ | |||
| static inline int get_se(GetBitContext *s, const char *file, const char *func, | |||
| int line) | |||
| { | |||
| int show= show_bits(s, 24); | |||
| int pos= get_bits_count(s); | |||
| int i= get_se_golomb(s); | |||
| @@ -147,11 +147,11 @@ int ff_h264_fill_default_ref_list(H264Context *h){ | |||
| } | |||
| #ifdef TRACE | |||
| for (i=0; i<h->ref_count[0]; i++) { | |||
| tprintf(h->s.avctx, "List0: %s fn:%d 0x%p\n", (h->default_ref_list[0][i].long_ref ? "LT" : "ST"), h->default_ref_list[0][i].pic_id, h->default_ref_list[0][i].data[0]); | |||
| tprintf(h->s.avctx, "List0: %s fn:%d 0x%p\n", (h->default_ref_list[0][i].long_ref ? "LT" : "ST"), h->default_ref_list[0][i].pic_id, h->default_ref_list[0][i].f.data[0]); | |||
| } | |||
| if(h->slice_type_nos==AV_PICTURE_TYPE_B){ | |||
| for (i=0; i<h->ref_count[1]; i++) { | |||
| tprintf(h->s.avctx, "List1: %s fn:%d 0x%p\n", (h->default_ref_list[1][i].long_ref ? "LT" : "ST"), h->default_ref_list[1][i].pic_id, h->default_ref_list[1][i].data[0]); | |||
| tprintf(h->s.avctx, "List1: %s fn:%d 0x%p\n", (h->default_ref_list[1][i].long_ref ? "LT" : "ST"), h->default_ref_list[1][i].pic_id, h->default_ref_list[1][i].f.data[0]); | |||
| } | |||
| } | |||
| #endif | |||
| @@ -21,11 +21,14 @@ | |||
| #include <opus.h> | |||
| #include <opus_multistream.h> | |||
| #include "libavutil/common.h" | |||
| #include "libavutil/avassert.h" | |||
| #include "libavutil/intreadwrite.h" | |||
| #include "avcodec.h" | |||
| #include "internal.h" | |||
| #include "vorbis.h" | |||
| #include "libavutil/avassert.h" | |||
| #include "libavutil/intreadwrite.h" | |||
| #include "mathops.h" | |||
| struct libopus_context { | |||
| OpusMSDecoder *dec; | |||
| @@ -36,7 +39,7 @@ struct libopus_context { | |||
| #endif | |||
| }; | |||
| static int ff_opus_error_to_averror(int err) | |||
| static int opus_error_to_averror(int err) | |||
| { | |||
| switch (err) { | |||
| case OPUS_BAD_ARG: return AVERROR(EINVAL); | |||
| @@ -50,40 +53,24 @@ static int ff_opus_error_to_averror(int err) | |||
| } | |||
| } | |||
| static inline void reorder(uint8_t *data, unsigned channels, unsigned bps, | |||
| unsigned samples, const uint8_t *map) | |||
| { | |||
| uint8_t tmp[8 * 4]; | |||
| unsigned i; | |||
| av_assert1(channels * bps <= sizeof(tmp)); | |||
| for (; samples > 0; samples--) { | |||
| for (i = 0; i < channels; i++) | |||
| memcpy(tmp + bps * i, data + bps * map[i], bps); | |||
| memcpy(data, tmp, bps * channels); | |||
| data += bps * channels; | |||
| } | |||
| } | |||
| #define OPUS_HEAD_SIZE 19 | |||
| static av_cold int libopus_dec_init(AVCodecContext *avc) | |||
| static av_cold int libopus_decode_init(AVCodecContext *avc) | |||
| { | |||
| struct libopus_context *opus = avc->priv_data; | |||
| int ret, channel_map = 0, gain_db = 0, nb_streams, nb_coupled; | |||
| uint8_t mapping_stereo[] = { 0, 1 }, *mapping; | |||
| uint8_t mapping_arr[8] = { 0, 1 }, *mapping; | |||
| avc->sample_rate = 48000; | |||
| avc->sample_fmt = avc->request_sample_fmt == AV_SAMPLE_FMT_FLT ? | |||
| AV_SAMPLE_FMT_FLT : AV_SAMPLE_FMT_S16; | |||
| avc->sample_rate = 48000; | |||
| avc->sample_fmt = avc->request_sample_fmt == AV_SAMPLE_FMT_FLT ? | |||
| AV_SAMPLE_FMT_FLT : AV_SAMPLE_FMT_S16; | |||
| avc->channel_layout = avc->channels > 8 ? 0 : | |||
| ff_vorbis_channel_layouts[avc->channels - 1]; | |||
| if (avc->extradata_size >= OPUS_HEAD_SIZE) { | |||
| opus->pre_skip = AV_RL16(avc->extradata + 10); | |||
| gain_db = AV_RL16(avc->extradata + 16); | |||
| channel_map = AV_RL8 (avc->extradata + 18); | |||
| gain_db -= (gain_db & 0x8000) << 1; /* signed */ | |||
| gain_db = sign_extend(AV_RL16(avc->extradata + 16), 16); | |||
| channel_map = AV_RL8 (avc->extradata + 18); | |||
| } | |||
| if (avc->extradata_size >= OPUS_HEAD_SIZE + 2 + avc->channels) { | |||
| nb_streams = avc->extradata[OPUS_HEAD_SIZE + 0]; | |||
| @@ -99,16 +86,26 @@ static av_cold int libopus_dec_init(AVCodecContext *avc) | |||
| } | |||
| nb_streams = 1; | |||
| nb_coupled = avc->channels > 1; | |||
| mapping = mapping_stereo; | |||
| mapping = mapping_arr; | |||
| } | |||
| if (avc->channels > 2 && avc->channels <= 8) { | |||
| const uint8_t *vorbis_offset = ff_vorbis_channel_layout_offsets[avc->channels - 1]; | |||
| int ch; | |||
| /* Remap channels from vorbis order to ffmpeg order */ | |||
| for (ch = 0; ch < avc->channels; ch++) | |||
| mapping_arr[ch] = mapping[vorbis_offset[ch]]; | |||
| mapping = mapping_arr; | |||
| } | |||
| opus->dec = opus_multistream_decoder_create( | |||
| avc->sample_rate, avc->channels, | |||
| nb_streams, nb_coupled, mapping, &ret); | |||
| opus->dec = opus_multistream_decoder_create(avc->sample_rate, avc->channels, | |||
| nb_streams, nb_coupled, | |||
| mapping, &ret); | |||
| if (!opus->dec) { | |||
| av_log(avc, AV_LOG_ERROR, "Unable to create decoder: %s\n", | |||
| opus_strerror(ret)); | |||
| return ff_opus_error_to_averror(ret); | |||
| return opus_error_to_averror(ret); | |||
| } | |||
| #ifdef OPUS_SET_GAIN | |||
| @@ -127,12 +124,13 @@ static av_cold int libopus_dec_init(AVCodecContext *avc) | |||
| #endif | |||
| avc->internal->skip_samples = opus->pre_skip; | |||
| avc->delay = 3840; /* Decoder delay (in samples) at 48kHz */ | |||
| avcodec_get_frame_defaults(&opus->frame); | |||
| avc->coded_frame = &opus->frame; | |||
| return 0; | |||
| } | |||
| static av_cold int libopus_dec_close(AVCodecContext *avc) | |||
| static av_cold int libopus_decode_close(AVCodecContext *avc) | |||
| { | |||
| struct libopus_context *opus = avc->priv_data; | |||
| @@ -140,10 +138,10 @@ static av_cold int libopus_dec_close(AVCodecContext *avc) | |||
| return 0; | |||
| } | |||
| #define MAX_FRAME_SIZE (960*6) | |||
| #define MAX_FRAME_SIZE (960 * 6) | |||
| static int libopus_dec_decode(AVCodecContext *avc, void *frame, | |||
| int *got_frame_ptr, AVPacket *pkt) | |||
| static int libopus_decode(AVCodecContext *avc, void *frame, | |||
| int *got_frame_ptr, AVPacket *pkt) | |||
| { | |||
| struct libopus_context *opus = avc->priv_data; | |||
| int ret, nb_samples; | |||
| @@ -155,25 +153,19 @@ static int libopus_dec_decode(AVCodecContext *avc, void *frame, | |||
| return ret; | |||
| } | |||
| nb_samples = avc->sample_fmt == AV_SAMPLE_FMT_S16 ? | |||
| opus_multistream_decode (opus->dec, pkt->data, pkt->size, | |||
| (void *)opus->frame.data[0], | |||
| opus->frame.nb_samples, 0) : | |||
| opus_multistream_decode_float(opus->dec, pkt->data, pkt->size, | |||
| (void *)opus->frame.data[0], | |||
| opus->frame.nb_samples, 0); | |||
| if (avc->sample_fmt == AV_SAMPLE_FMT_S16) | |||
| nb_samples = opus_multistream_decode(opus->dec, pkt->data, pkt->size, | |||
| (opus_int16 *)opus->frame.data[0], | |||
| opus->frame.nb_samples, 0); | |||
| else | |||
| nb_samples = opus_multistream_decode_float(opus->dec, pkt->data, pkt->size, | |||
| (float *)opus->frame.data[0], | |||
| opus->frame.nb_samples, 0); | |||
| if (nb_samples < 0) { | |||
| av_log(avc, AV_LOG_ERROR, "Decoding error: %s\n", | |||
| opus_strerror(nb_samples)); | |||
| return ff_opus_error_to_averror(nb_samples); | |||
| } | |||
| if (avc->channels > 3 && avc->channels <= 8) { | |||
| const uint8_t *m = ff_vorbis_channel_layout_offsets[avc->channels - 1]; | |||
| if (avc->sample_fmt == AV_SAMPLE_FMT_S16) | |||
| reorder(opus->frame.data[0], avc->channels, 2, nb_samples, m); | |||
| else | |||
| reorder(opus->frame.data[0], avc->channels, 4, nb_samples, m); | |||
| return opus_error_to_averror(nb_samples); | |||
| } | |||
| #ifndef OPUS_SET_GAIN | |||
| @@ -197,7 +189,7 @@ static int libopus_dec_decode(AVCodecContext *avc, void *frame, | |||
| return pkt->size; | |||
| } | |||
| static void libopus_dec_flush(AVCodecContext *avc) | |||
| static void libopus_flush(AVCodecContext *avc) | |||
| { | |||
| struct libopus_context *opus = avc->priv_data; | |||
| @@ -212,10 +204,13 @@ AVCodec ff_libopus_decoder = { | |||
| .type = AVMEDIA_TYPE_AUDIO, | |||
| .id = AV_CODEC_ID_OPUS, | |||
| .priv_data_size = sizeof(struct libopus_context), | |||
| .init = libopus_dec_init, | |||
| .close = libopus_dec_close, | |||
| .decode = libopus_dec_decode, | |||
| .flush = libopus_dec_flush, | |||
| .init = libopus_decode_init, | |||
| .close = libopus_decode_close, | |||
| .decode = libopus_decode, | |||
| .flush = libopus_flush, | |||
| .capabilities = CODEC_CAP_DR1, | |||
| .long_name = NULL_IF_CONFIG_SMALL("libopus Opus"), | |||
| .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT, | |||
| AV_SAMPLE_FMT_S16, | |||
| AV_SAMPLE_FMT_NONE }, | |||
| }; | |||
| @@ -5704,6 +5704,7 @@ AVCodec ff_vc1_decoder = { | |||
| .init = vc1_decode_init, | |||
| .close = ff_vc1_decode_end, | |||
| .decode = vc1_decode_frame, | |||
| .flush = ff_mpeg_flush, | |||
| .capabilities = CODEC_CAP_DR1 | CODEC_CAP_DELAY, | |||
| .long_name = NULL_IF_CONFIG_SMALL("SMPTE VC-1"), | |||
| .pix_fmts = ff_hwaccel_pixfmt_list_420, | |||
| @@ -5719,6 +5720,7 @@ AVCodec ff_wmv3_decoder = { | |||
| .init = vc1_decode_init, | |||
| .close = ff_vc1_decode_end, | |||
| .decode = vc1_decode_frame, | |||
| .flush = ff_mpeg_flush, | |||
| .capabilities = CODEC_CAP_DR1 | CODEC_CAP_DELAY, | |||
| .long_name = NULL_IF_CONFIG_SMALL("Windows Media Video 9"), | |||
| .pix_fmts = ff_hwaccel_pixfmt_list_420, | |||
| @@ -27,7 +27,7 @@ | |||
| */ | |||
| #define LIBAVCODEC_VERSION_MAJOR 54 | |||
| #define LIBAVCODEC_VERSION_MINOR 60 | |||
| #define LIBAVCODEC_VERSION_MINOR 61 | |||
| #define LIBAVCODEC_VERSION_MICRO 100 | |||
| #define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \ | |||
| @@ -32,8 +32,11 @@ | |||
| /* XXX: POST protocol is not completely implemented because ffmpeg uses | |||
| only a subset of it. */ | |||
| /* used for protocol handling */ | |||
| #define BUFFER_SIZE 4096 | |||
| /* The IO buffer size is unrelated to the max URL size in itself, but needs | |||
| * to be large enough to fit the full request headers (including long | |||
| * path names). | |||
| */ | |||
| #define BUFFER_SIZE MAX_URL_SIZE | |||
| #define MAX_REDIRECTS 8 | |||
| typedef struct { | |||
| @@ -101,8 +104,8 @@ static int http_open_cnx(URLContext *h) | |||
| const char *path, *proxy_path, *lower_proto = "tcp", *local_path; | |||
| char hostname[1024], hoststr[1024], proto[10]; | |||
| char auth[1024], proxyauth[1024] = ""; | |||
| char path1[1024]; | |||
| char buf[1024], urlbuf[1024]; | |||
| char path1[MAX_URL_SIZE]; | |||
| char buf[1024], urlbuf[MAX_URL_SIZE]; | |||
| int port, use_proxy, err, location_changed = 0, redirects = 0, attempts = 0; | |||
| HTTPAuthType cur_auth_type, cur_proxy_auth_type; | |||
| HTTPContext *s = h->priv_data; | |||
| @@ -352,7 +355,7 @@ static inline int has_header(const char *str, const char *header) | |||
| static int http_read_header(URLContext *h, int *new_location) | |||
| { | |||
| HTTPContext *s = h->priv_data; | |||
| char line[1024]; | |||
| char line[MAX_URL_SIZE]; | |||
| int err = 0; | |||
| s->chunksize = -1; | |||
| @@ -266,13 +266,17 @@ static void put_s(AVIOContext *bc, int64_t val){ | |||
| } | |||
| #ifdef TRACE | |||
| static inline void ff_put_v_trace(AVIOContext *bc, uint64_t v, char *file, char *func, int line){ | |||
| static inline void ff_put_v_trace(AVIOContext *bc, uint64_t v, const char *file, | |||
| const char *func, int line) | |||
| { | |||
| av_log(NULL, AV_LOG_DEBUG, "ff_put_v %5"PRId64" / %"PRIX64" in %s %s:%d\n", v, v, file, func, line); | |||
| ff_put_v(bc, v); | |||
| } | |||
| static inline void put_s_trace(AVIOContext *bc, int64_t v, char *file, char *func, int line){ | |||
| static inline void put_s_trace(AVIOContext *bc, int64_t v, const char *file, | |||
| const char *func, int line) | |||
| { | |||
| av_log(NULL, AV_LOG_DEBUG, "put_s %5"PRId64" / %"PRIX64" in %s %s:%d\n", v, v, file, func, line); | |||
| put_s(bc, v); | |||