diff options
author | Michael Niedermayer <michaelni@gmx.at> | 2011-12-03 05:08:55 +0400 |
---|---|---|
committer | Michael Niedermayer <michaelni@gmx.at> | 2011-12-03 06:00:30 +0400 |
commit | e4de71677f3adeac0f74b89ac8df5d417364df2c (patch) | |
tree | 4792dd8d85d24f0f4eaddabb65f6044727907daa /libavcodec/aacdec.c | |
parent | 12804348f5babf56a315fa01751eea1ffdddf98a (diff) | |
parent | d268b79e3436107c11ee8bcdf9f3645368bb3fcd (diff) |
Merge remote-tracking branch 'qatar/master'
* qatar/master:
aac_latm: reconfigure decoder on audio specific config changes
latmdec: fix audio specific config parsing
Add avcodec_decode_audio4().
avcodec: change number of plane pointers from 4 to 8 at next major bump.
Update developers documentation with coding conventions.
svq1dec: avoid undefined get_bits(0) call
ARM: h264dsp_neon cosmetics
ARM: make some NEON macros reusable
Do not memcpy raw video frames when using null muxer
fate: update asf seektest
vp8: flush buffers on size changes.
doc: improve general documentation for MacOSX
asf: use packet dts as approximation of pts
asf: do not call av_read_frame
rtsp: Initialize the media_type_mask in the rtp guessing demuxer
Cleaned up alacenc.c
Conflicts:
doc/APIchanges
doc/developer.texi
libavcodec/8svx.c
libavcodec/aacdec.c
libavcodec/ac3dec.c
libavcodec/avcodec.h
libavcodec/nellymoserdec.c
libavcodec/tta.c
libavcodec/utils.c
libavcodec/version.h
libavcodec/wmadec.c
libavformat/asfdec.c
tests/ref/seek/lavf_asf
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavcodec/aacdec.c')
-rw-r--r-- | libavcodec/aacdec.c | 106 |
1 files changed, 61 insertions, 45 deletions
diff --git a/libavcodec/aacdec.c b/libavcodec/aacdec.c index 5a2b230d24..a046d991e6 100644 --- a/libavcodec/aacdec.c +++ b/libavcodec/aacdec.c @@ -471,15 +471,17 @@ static int decode_ga_specific_config(AACContext *ac, AVCodecContext *avctx, * @param ac pointer to AACContext, may be null * @param avctx pointer to AVCCodecContext, used for logging * @param m4ac pointer to MPEG4AudioConfig, used for parsing - * @param data pointer to AVCodecContext extradata - * @param data_size size of AVCCodecContext extradata + * @param data pointer to buffer holding an audio specific config + * @param bit_size size of audio specific config or data in bits + * @param sync_extension look for an appended sync extension * * @return Returns error status or number of consumed bits. <0 - error */ static int decode_audio_specific_config(AACContext *ac, AVCodecContext *avctx, MPEG4AudioConfig *m4ac, - const uint8_t *data, int data_size, int asclen) + const uint8_t *data, int bit_size, + int sync_extension) { GetBitContext gb; int i; @@ -489,9 +491,9 @@ static int decode_audio_specific_config(AACContext *ac, av_dlog(avctx, "%02x ", avctx->extradata[i]); av_dlog(avctx, "\n"); - init_get_bits(&gb, data, data_size * 8); + init_get_bits(&gb, data, bit_size); - if ((i = avpriv_mpeg4audio_get_config(m4ac, data, asclen/8)) < 0) + if ((i = avpriv_mpeg4audio_get_config(m4ac, data, bit_size, sync_extension)) < 0) return -1; if (m4ac->sampling_index > 12) { av_log(avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", m4ac->sampling_index); @@ -591,7 +593,7 @@ static av_cold int aac_decode_init(AVCodecContext *avctx) if (avctx->extradata_size > 0) { if (decode_audio_specific_config(ac, ac->avctx, &ac->m4ac, avctx->extradata, - avctx->extradata_size, 8*avctx->extradata_size) < 0) + avctx->extradata_size*8, 1) < 0) return -1; } else { int sr, i; @@ -665,6 +667,9 @@ static av_cold int aac_decode_init(AVCodecContext *avctx) cbrt_tableinit(); + avcodec_get_frame_defaults(&ac->frame); + avctx->coded_frame = &ac->frame; + return 0; } @@ -2132,12 +2137,12 @@ static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb) } static int aac_decode_frame_int(AVCodecContext *avctx, void *data, - int *data_size, GetBitContext *gb) + int *got_frame_ptr, GetBitContext *gb) { AACContext *ac = avctx->priv_data; ChannelElement *che = NULL, *che_prev = NULL; enum RawDataBlockType elem_type, elem_type_prev = TYPE_END; - int err, elem_id, data_size_tmp; + int err, elem_id; int samples = 0, multiplier, audio_found = 0; if (show_bits(gb, 12) == 0xfff) { @@ -2250,24 +2255,26 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data, avctx->frame_size = samples; } - data_size_tmp = samples * avctx->channels * - av_get_bytes_per_sample(avctx->sample_fmt); - if (*data_size < data_size_tmp) { - av_log(avctx, AV_LOG_ERROR, - "Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n", - *data_size, data_size_tmp); - return -1; - } - *data_size = data_size_tmp; - if (samples) { + /* get output buffer */ + ac->frame.nb_samples = samples; + if ((err = avctx->get_buffer(avctx, &ac->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return err; + } + if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT) - ac->fmt_conv.float_interleave(data, (const float **)ac->output_data, + ac->fmt_conv.float_interleave((float *)ac->frame.data[0], + (const float **)ac->output_data, samples, avctx->channels); else - ac->fmt_conv.float_to_int16_interleave(data, (const float **)ac->output_data, + ac->fmt_conv.float_to_int16_interleave((int16_t *)ac->frame.data[0], + (const float **)ac->output_data, samples, avctx->channels); + + *(AVFrame *)data = ac->frame; } + *got_frame_ptr = !!samples; if (ac->output_configured && audio_found) ac->output_configured = OC_LOCKED; @@ -2276,7 +2283,7 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data, } static int aac_decode_frame(AVCodecContext *avctx, void *data, - int *data_size, AVPacket *avpkt) + int *got_frame_ptr, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; @@ -2287,7 +2294,7 @@ static int aac_decode_frame(AVCodecContext *avctx, void *data, init_get_bits(&gb, buf, buf_size * 8); - if ((err = aac_decode_frame_int(avctx, data, data_size, &gb)) < 0) + if ((err = aac_decode_frame_int(avctx, data, got_frame_ptr, &gb)) < 0) return err; buf_consumed = (get_bits_count(&gb) + 7) >> 3; @@ -2340,30 +2347,40 @@ static inline uint32_t latm_get_value(GetBitContext *b) static int latm_decode_audio_specific_config(struct LATMContext *latmctx, GetBitContext *gb, int asclen) { - AVCodecContext *avctx = latmctx->aac_ctx.avctx; - AACContext *ac= &latmctx->aac_ctx; - MPEG4AudioConfig m4ac=ac->m4ac; - int config_start_bit = get_bits_count(gb); - int bits_consumed, esize; + AACContext *ac = &latmctx->aac_ctx; + AVCodecContext *avctx = ac->avctx; + MPEG4AudioConfig m4ac = {0}; + int config_start_bit = get_bits_count(gb); + int sync_extension = 0; + int bits_consumed, esize; + + if (asclen) { + sync_extension = 1; + asclen = FFMIN(asclen, get_bits_left(gb)); + } else + asclen = get_bits_left(gb); if (config_start_bit % 8) { av_log_missing_feature(latmctx->aac_ctx.avctx, "audio specific " "config not byte aligned.\n", 1); return AVERROR_INVALIDDATA; - } else { - bits_consumed = - decode_audio_specific_config(ac, avctx, &m4ac, + } + bits_consumed = decode_audio_specific_config(NULL, avctx, &m4ac, gb->buffer + (config_start_bit / 8), - get_bits_left(gb) / 8, asclen); + asclen, sync_extension); - if (bits_consumed < 0) - return AVERROR_INVALIDDATA; - if(ac->m4ac.sample_rate != m4ac.sample_rate || m4ac.chan_config != ac->m4ac.chan_config) - ac->m4ac= m4ac; + if (bits_consumed < 0) + return AVERROR_INVALIDDATA; + + if (ac->m4ac.sample_rate != m4ac.sample_rate || + ac->m4ac.chan_config != m4ac.chan_config) { + + av_log(avctx, AV_LOG_INFO, "audio config changed\n"); + latmctx->initialized = 0; esize = (bits_consumed+7) / 8; - if (avctx->extradata_size <= esize) { + if (avctx->extradata_size < esize) { av_free(avctx->extradata); avctx->extradata = av_malloc(esize + FF_INPUT_BUFFER_PADDING_SIZE); if (!avctx->extradata) @@ -2373,9 +2390,8 @@ static int latm_decode_audio_specific_config(struct LATMContext *latmctx, avctx->extradata_size = esize; memcpy(avctx->extradata, gb->buffer + (config_start_bit/8), esize); memset(avctx->extradata+esize, 0, FF_INPUT_BUFFER_PADDING_SIZE); - - skip_bits_long(gb, bits_consumed); } + skip_bits_long(gb, bits_consumed); return bits_consumed; } @@ -2512,8 +2528,8 @@ static int read_audio_mux_element(struct LATMContext *latmctx, } -static int latm_decode_frame(AVCodecContext *avctx, void *out, int *out_size, - AVPacket *avpkt) +static int latm_decode_frame(AVCodecContext *avctx, void *out, + int *got_frame_ptr, AVPacket *avpkt) { struct LATMContext *latmctx = avctx->priv_data; int muxlength, err; @@ -2535,12 +2551,12 @@ static int latm_decode_frame(AVCodecContext *avctx, void *out, int *out_size, if (!latmctx->initialized) { if (!avctx->extradata) { - *out_size = 0; + *got_frame_ptr = 0; return avpkt->size; } else { if ((err = decode_audio_specific_config( &latmctx->aac_ctx, avctx, &latmctx->aac_ctx.m4ac, - avctx->extradata, avctx->extradata_size, 8*avctx->extradata_size)) < 0) + avctx->extradata, avctx->extradata_size*8, 1)) < 0) return err; latmctx->initialized = 1; } @@ -2553,7 +2569,7 @@ static int latm_decode_frame(AVCodecContext *avctx, void *out, int *out_size, return AVERROR_INVALIDDATA; } - if ((err = aac_decode_frame_int(avctx, out, out_size, &gb)) < 0) + if ((err = aac_decode_frame_int(avctx, out, got_frame_ptr, &gb)) < 0) return err; return muxlength; @@ -2583,7 +2599,7 @@ AVCodec ff_aac_decoder = { .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE }, - .capabilities = CODEC_CAP_CHANNEL_CONF, + .capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1, .channel_layouts = aac_channel_layout, }; @@ -2604,7 +2620,7 @@ AVCodec ff_aac_latm_decoder = { .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE }, - .capabilities = CODEC_CAP_CHANNEL_CONF, + .capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1, .channel_layouts = aac_channel_layout, .flush = flush, }; |