diff options
author | Michael Niedermayer <michaelni@gmx.at> | 2012-11-02 17:15:28 +0400 |
---|---|---|
committer | Michael Niedermayer <michaelni@gmx.at> | 2012-11-02 17:20:33 +0400 |
commit | 6788350281c418f0f395a8279eee82f7abe7c63b (patch) | |
tree | 69cd76f699eff929f5b13f76b42eabc7f25f9355 /libavcodec/flacdec.c | |
parent | 00aa7fa786e41b5fc8404732453869aa3c14e33a (diff) | |
parent | 50a65e7a540ce6747f81d6dbf6a602ad35be77ff (diff) |
Merge commit '50a65e7a540ce6747f81d6dbf6a602ad35be77ff'
* commit '50a65e7a540ce6747f81d6dbf6a602ad35be77ff': (24 commits)
vmdaudio: set channel layout
twinvq: validate sample rate code
twinvq: set channel layout
twinvq: validate that channels is not <= 0
truespeech: set channel layout
sipr: set channel layout
shorten: validate that the channel count in the header is not <= 0
ra288dec: set channel layout
ra144dec: set channel layout
qdm2: remove unneeded checks for channel count
qdm2: make sure channels is not <= 0 and set channel layout
qcelpdec: set channel layout
nellymoserdec: set channels to 1
libopencore-amr: set channel layout for amr-nb or if not set by the user
libilbc: set channel layout
dpcm: use AVCodecContext.channels instead of keeping a private copy
imc: set channels to 1 instead of validating it
gsmdec: always set channel layout and sample rate at initialization
libgsmdec: always set channel layout and sample rate at initialization
g726dec: do not validate sample rate
...
Conflicts:
libavcodec/dpcm.c
libavcodec/qdm2.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavcodec/flacdec.c')
-rw-r--r-- | libavcodec/flacdec.c | 74 |
1 files changed, 38 insertions, 36 deletions
diff --git a/libavcodec/flacdec.c b/libavcodec/flacdec.c index 992127713c..05793c961d 100644 --- a/libavcodec/flacdec.c +++ b/libavcodec/flacdec.c @@ -58,20 +58,13 @@ typedef struct FLACContext { int got_streaminfo; ///< indicates if the STREAMINFO has been read int32_t *decoded[FLAC_MAX_CHANNELS]; ///< decoded samples + uint8_t *decoded_buffer; + unsigned int decoded_buffer_size; FLACDSPContext dsp; } FLACContext; -static const int64_t flac_channel_layouts[6] = { - AV_CH_LAYOUT_MONO, - AV_CH_LAYOUT_STEREO, - AV_CH_LAYOUT_SURROUND, - AV_CH_LAYOUT_QUAD, - AV_CH_LAYOUT_5POINT0, - AV_CH_LAYOUT_5POINT1 -}; - -static void allocate_buffers(FLACContext *s); +static int allocate_buffers(FLACContext *s); static void flac_set_bps(FLACContext *s) { @@ -99,6 +92,7 @@ static av_cold int flac_decode_init(AVCodecContext *avctx) { enum FLACExtradataFormat format; uint8_t *streaminfo; + int ret; FLACContext *s = avctx->priv_data; s->avctx = avctx; @@ -112,7 +106,9 @@ static av_cold int flac_decode_init(AVCodecContext *avctx) /* initialize based on the demuxer-supplied streamdata header */ avpriv_flac_parse_streaminfo(avctx, (FLACStreaminfo *)s, streaminfo); - allocate_buffers(s); + ret = allocate_buffers(s); + if (ret < 0) + return ret; flac_set_bps(s); ff_flacdsp_init(&s->dsp, avctx->sample_fmt, s->bps); s->got_streaminfo = 1; @@ -120,9 +116,6 @@ static av_cold int flac_decode_init(AVCodecContext *avctx) avcodec_get_frame_defaults(&s->frame); avctx->coded_frame = &s->frame; - if (avctx->channels <= FF_ARRAY_ELEMS(flac_channel_layouts)) - avctx->channel_layout = flac_channel_layouts[avctx->channels - 1]; - return 0; } @@ -135,15 +128,24 @@ static void dump_headers(AVCodecContext *avctx, FLACStreaminfo *s) av_log(avctx, AV_LOG_DEBUG, " Bits: %d\n", s->bps); } -static void allocate_buffers(FLACContext *s) +static int allocate_buffers(FLACContext *s) { - int i; + int buf_size; av_assert0(s->max_blocksize); - for (i = 0; i < s->channels; i++) { - s->decoded[i] = av_malloc(sizeof(int32_t)*s->max_blocksize); - } + buf_size = av_samples_get_buffer_size(NULL, s->channels, s->max_blocksize, + AV_SAMPLE_FMT_S32P, 0); + if (buf_size < 0) + return buf_size; + + av_fast_malloc(&s->decoded_buffer, &s->decoded_buffer_size, buf_size); + if (!s->decoded_buffer) + return AVERROR(ENOMEM); + + return av_samples_fill_arrays((uint8_t **)s->decoded, NULL, + s->decoded_buffer, s->channels, + s->max_blocksize, AV_SAMPLE_FMT_S32P, 0); } /** @@ -155,7 +157,7 @@ static void allocate_buffers(FLACContext *s) */ static int parse_streaminfo(FLACContext *s, const uint8_t *buf, int buf_size) { - int metadata_type, metadata_size; + int metadata_type, metadata_size, ret; if (buf_size < FLAC_STREAMINFO_SIZE+8) { /* need more data */ @@ -167,7 +169,9 @@ static int parse_streaminfo(FLACContext *s, const uint8_t *buf, int buf_size) return AVERROR_INVALIDDATA; } avpriv_flac_parse_streaminfo(s->avctx, (FLACStreaminfo *)s, &buf[8]); - allocate_buffers(s); + ret = allocate_buffers(s); + if (ret < 0) + return ret; flac_set_bps(s); ff_flacdsp_init(&s->dsp, s->avctx->sample_fmt, s->bps); s->got_streaminfo = 1; @@ -403,7 +407,7 @@ static inline int decode_subframe(FLACContext *s, int channel) static int decode_frame(FLACContext *s) { - int i; + int i, ret; GetBitContext *gb = &s->gb; FLACFrameInfo fi; @@ -412,12 +416,15 @@ static int decode_frame(FLACContext *s) return -1; } - if (s->channels && fi.channels != s->channels) { - av_log(s->avctx, AV_LOG_ERROR, "switching channel layout mid-stream " - "is not supported\n"); - return -1; + if (s->channels && fi.channels != s->channels && s->got_streaminfo) { + s->channels = s->avctx->channels = fi.channels; + ff_flac_set_channel_layout(s->avctx); + ret = allocate_buffers(s); + if (ret < 0) + return ret; } s->channels = s->avctx->channels = fi.channels; + ff_flac_set_channel_layout(s->avctx); s->ch_mode = fi.ch_mode; if (!s->bps && !fi.bps) { @@ -451,16 +458,14 @@ static int decode_frame(FLACContext *s) " or frame header\n"); return -1; } - if (fi.samplerate == 0) { + if (fi.samplerate == 0) fi.samplerate = s->samplerate; - } else if (s->samplerate && fi.samplerate != s->samplerate) { - av_log(s->avctx, AV_LOG_WARNING, "sample rate changed from %d to %d\n", - s->samplerate, fi.samplerate); - } s->samplerate = s->avctx->sample_rate = fi.samplerate; if (!s->got_streaminfo) { - allocate_buffers(s); + ret = allocate_buffers(s); + if (ret < 0) + return ret; ff_flacdsp_init(&s->dsp, s->avctx->sample_fmt, s->bps); s->got_streaminfo = 1; dump_headers(s->avctx, (FLACStreaminfo *)s); @@ -550,11 +555,8 @@ static int flac_decode_frame(AVCodecContext *avctx, void *data, static av_cold int flac_decode_close(AVCodecContext *avctx) { FLACContext *s = avctx->priv_data; - int i; - for (i = 0; i < s->channels; i++) { - av_freep(&s->decoded[i]); - } + av_freep(&s->decoded_buffer); return 0; } |