From ba7d6e798eb11b463852479a5e09a68926e82af6 Mon Sep 17 00:00:00 2001 From: Stefano Sabatini Date: Wed, 3 Nov 2010 20:19:34 +0000 Subject: [PATCH] Remove usage of deprecated libavcodec/audioconvert.h functions. Originally committed as revision 25668 to svn://svn.ffmpeg.org/ffmpeg/trunk --- ffmpeg.c | 17 +++++++++-------- ffplay.c | 7 ++++--- libavcodec/alsdec.c | 5 +++-- libavcodec/resample.c | 9 +++++---- libavcodec/utils.c | 3 ++- libavfilter/defaults.c | 3 ++- libavformat/matroskaenc.c | 3 ++- 7 files changed, 27 insertions(+), 20 deletions(-) diff --git a/ffmpeg.c b/ffmpeg.c index 4d1b75416c..ad3f670e10 100644 --- a/ffmpeg.c +++ b/ffmpeg.c @@ -37,6 +37,7 @@ #include "libavcodec/opt.h" #include "libavcodec/audioconvert.h" #include "libavcore/parseutils.h" +#include "libavcore/samplefmt.h" #include "libavutil/colorspace.h" #include "libavutil/fifo.h" #include "libavutil/intreadwrite.h" @@ -769,8 +770,8 @@ static void do_audio_out(AVFormatContext *s, int size_out, frame_bytes, ret; AVCodecContext *enc= ost->st->codec; AVCodecContext *dec= ist->st->codec; - int osize= av_get_bits_per_sample_format(enc->sample_fmt)/8; - int isize= av_get_bits_per_sample_format(dec->sample_fmt)/8; + int osize= av_get_bits_per_sample_fmt(enc->sample_fmt)/8; + int isize= av_get_bits_per_sample_fmt(dec->sample_fmt)/8; const int coded_bps = av_get_bits_per_sample(enc->codec->id); need_realloc: @@ -824,8 +825,8 @@ need_realloc: dec->sample_fmt, 1, NULL, 0); if (!ost->reformat_ctx) { fprintf(stderr, "Cannot convert %s sample format to %s sample format\n", - avcodec_get_sample_fmt_name(dec->sample_fmt), - avcodec_get_sample_fmt_name(enc->sample_fmt)); + av_get_sample_fmt_name(dec->sample_fmt), + av_get_sample_fmt_name(enc->sample_fmt)); ffmpeg_exit(1); } ost->reformat_pair=MAKE_SFMT_PAIR(enc->sample_fmt,dec->sample_fmt); @@ -1443,7 +1444,7 @@ static int output_packet(AVInputStream *ist, int ist_index, #endif AVPacket avpkt; - int bps = av_get_bits_per_sample_format(ist->st->codec->sample_fmt)>>3; + int bps = av_get_bits_per_sample_fmt(ist->st->codec->sample_fmt)>>3; if(ist->next_pts == AV_NOPTS_VALUE) ist->next_pts= ist->pts; @@ -1760,7 +1761,7 @@ static int output_packet(AVInputStream *ist, int ist_index, ret = 0; /* encode any samples remaining in fifo */ if (fifo_bytes > 0) { - int osize = av_get_bits_per_sample_format(enc->sample_fmt) >> 3; + int osize = av_get_bits_per_sample_fmt(enc->sample_fmt) >> 3; int fs_tmp = enc->frame_size; av_fifo_generic_read(ost->fifo, audio_buf, fifo_bytes, NULL); @@ -2817,9 +2818,9 @@ static int opt_thread_count(const char *opt, const char *arg) static void opt_audio_sample_fmt(const char *arg) { if (strcmp(arg, "list")) - audio_sample_fmt = avcodec_get_sample_fmt(arg); + audio_sample_fmt = av_get_sample_fmt(arg); else { - list_fmts(avcodec_sample_fmt_string, SAMPLE_FMT_NB); + list_fmts(av_get_sample_fmt_string, SAMPLE_FMT_NB); ffmpeg_exit(0); } } diff --git a/ffplay.c b/ffplay.c index eecf16a14f..0563e96edc 100644 --- a/ffplay.c +++ b/ffplay.c @@ -30,6 +30,7 @@ #include "libavutil/pixdesc.h" #include "libavcore/imgutils.h" #include "libavcore/parseutils.h" +#include "libavcore/samplefmt.h" #include "libavformat/avformat.h" #include "libavdevice/avdevice.h" #include "libswscale/swscale.h" @@ -2099,8 +2100,8 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr) dec->sample_fmt, 1, NULL, 0); if (!is->reformat_ctx) { fprintf(stderr, "Cannot convert %s sample format to %s sample format\n", - avcodec_get_sample_fmt_name(dec->sample_fmt), - avcodec_get_sample_fmt_name(SAMPLE_FMT_S16)); + av_get_sample_fmt_name(dec->sample_fmt), + av_get_sample_fmt_name(SAMPLE_FMT_S16)); break; } is->audio_src_fmt= dec->sample_fmt; @@ -2109,7 +2110,7 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr) if (is->reformat_ctx) { const void *ibuf[6]= {is->audio_buf1}; void *obuf[6]= {is->audio_buf2}; - int istride[6]= {av_get_bits_per_sample_format(dec->sample_fmt)/8}; + int istride[6]= {av_get_bits_per_sample_fmt(dec->sample_fmt)/8}; int ostride[6]= {2}; int len= data_size/istride[0]; if (av_audio_convert(is->reformat_ctx, obuf, ostride, ibuf, istride, len)<0) { diff --git a/libavcodec/alsdec.c b/libavcodec/alsdec.c index 352506b159..6993bb98a6 100644 --- a/libavcodec/alsdec.c +++ b/libavcodec/alsdec.c @@ -36,6 +36,7 @@ #include "bytestream.h" #include "bgmc.h" #include "dsputil.h" +#include "libavcore/samplefmt.h" #include "libavutil/crc.h" #include @@ -1426,7 +1427,7 @@ static int decode_frame(AVCodecContext *avctx, // check for size of decoded data size = ctx->cur_frame_length * avctx->channels * - (av_get_bits_per_sample_format(avctx->sample_fmt) >> 3); + (av_get_bits_per_sample_fmt(avctx->sample_fmt) >> 3); if (size > *data_size) { av_log(avctx, AV_LOG_ERROR, "Decoded data exceeds buffer size.\n"); @@ -1679,7 +1680,7 @@ static av_cold int decode_init(AVCodecContext *avctx) ctx->crc_buffer = av_malloc(sizeof(*ctx->crc_buffer) * ctx->cur_frame_length * avctx->channels * - (av_get_bits_per_sample_format(avctx->sample_fmt) >> 3)); + (av_get_bits_per_sample_fmt(avctx->sample_fmt) >> 3)); if (!ctx->crc_buffer) { av_log(avctx, AV_LOG_ERROR, "Allocating buffer memory failed.\n"); decode_end(avctx); diff --git a/libavcodec/resample.c b/libavcodec/resample.c index 222b28ce83..89e2d71e53 100644 --- a/libavcodec/resample.c +++ b/libavcodec/resample.c @@ -27,6 +27,7 @@ #include "avcodec.h" #include "audioconvert.h" #include "libavutil/opt.h" +#include "libavcore/samplefmt.h" struct AVResampleContext; @@ -174,15 +175,15 @@ ReSampleContext *av_audio_resample_init(int output_channels, int input_channels, s->sample_fmt [0] = sample_fmt_in; s->sample_fmt [1] = sample_fmt_out; - s->sample_size[0] = av_get_bits_per_sample_format(s->sample_fmt[0])>>3; - s->sample_size[1] = av_get_bits_per_sample_format(s->sample_fmt[1])>>3; + s->sample_size[0] = av_get_bits_per_sample_fmt(s->sample_fmt[0])>>3; + s->sample_size[1] = av_get_bits_per_sample_fmt(s->sample_fmt[1])>>3; if (s->sample_fmt[0] != SAMPLE_FMT_S16) { if (!(s->convert_ctx[0] = av_audio_convert_alloc(SAMPLE_FMT_S16, 1, s->sample_fmt[0], 1, NULL, 0))) { av_log(s, AV_LOG_ERROR, "Cannot convert %s sample format to s16 sample format\n", - avcodec_get_sample_fmt_name(s->sample_fmt[0])); + av_get_sample_fmt_name(s->sample_fmt[0])); av_free(s); return NULL; } @@ -193,7 +194,7 @@ ReSampleContext *av_audio_resample_init(int output_channels, int input_channels, SAMPLE_FMT_S16, 1, NULL, 0))) { av_log(s, AV_LOG_ERROR, "Cannot convert s16 sample format to %s sample format\n", - avcodec_get_sample_fmt_name(s->sample_fmt[1])); + av_get_sample_fmt_name(s->sample_fmt[1])); av_audio_convert_free(s->convert_ctx[0]); av_free(s); return NULL; diff --git a/libavcodec/utils.c b/libavcodec/utils.c index 8169b374b5..4d13e0aa6e 100644 --- a/libavcodec/utils.c +++ b/libavcodec/utils.c @@ -30,6 +30,7 @@ #include "libavutil/crc.h" #include "libavutil/pixdesc.h" #include "libavcore/imgutils.h" +#include "libavcore/samplefmt.h" #include "avcodec.h" #include "dsputil.h" #include "libavutil/opt.h" @@ -923,7 +924,7 @@ void avcodec_string(char *buf, int buf_size, AVCodecContext *enc, int encode) avcodec_get_channel_layout_string(buf + strlen(buf), buf_size - strlen(buf), enc->channels, enc->channel_layout); if (enc->sample_fmt != SAMPLE_FMT_NONE) { snprintf(buf + strlen(buf), buf_size - strlen(buf), - ", %s", avcodec_get_sample_fmt_name(enc->sample_fmt)); + ", %s", av_get_sample_fmt_name(enc->sample_fmt)); } break; case AVMEDIA_TYPE_DATA: diff --git a/libavfilter/defaults.c b/libavfilter/defaults.c index 1d748c2855..5462b1a34a 100644 --- a/libavfilter/defaults.c +++ b/libavfilter/defaults.c @@ -20,6 +20,7 @@ */ #include "libavcore/imgutils.h" +#include "libavcore/samplefmt.h" #include "libavcodec/audioconvert.h" #include "avfilter.h" @@ -109,7 +110,7 @@ AVFilterBufferRef *avfilter_default_get_audio_buffer(AVFilterLink *link, int per samples->refcount = 1; samples->free = avfilter_default_free_buffer; - sample_size = av_get_bits_per_sample_format(sample_fmt) >>3; + sample_size = av_get_bits_per_sample_fmt(sample_fmt) >>3; chans_nb = avcodec_channel_layout_num_channels(channel_layout); per_channel_size = size/chans_nb; diff --git a/libavformat/matroskaenc.c b/libavformat/matroskaenc.c index 80bb08ef4f..1430228c00 100644 --- a/libavformat/matroskaenc.c +++ b/libavformat/matroskaenc.c @@ -26,6 +26,7 @@ #include "avc.h" #include "flacenc.h" #include "avlanguage.h" +#include "libavcore/samplefmt.h" #include "libavutil/intreadwrite.h" #include "libavutil/random_seed.h" #include "libavutil/lfg.h" @@ -540,7 +541,7 @@ static int mkv_write_tracks(AVFormatContext *s) AVMetadataTag *tag; if (!bit_depth) - bit_depth = av_get_bits_per_sample_format(codec->sample_fmt); + bit_depth = av_get_bits_per_sample_fmt(codec->sample_fmt); if (codec->codec_id == CODEC_ID_AAC) get_aac_sample_rates(s, codec, &sample_rate, &output_sample_rate); -- GitLab