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author | Anton Khirnov <anton@khirnov.net> | 2013-02-23 08:20:12 +0100 |
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committer | Anton Khirnov <anton@khirnov.net> | 2013-03-09 08:36:40 +0100 |
commit | 5d606863c353553488e88dfd89ea5571268d9dca (patch) | |
tree | b1105afa8f020bca609a15eafc669876139429a8 | |
parent | 6327c10702922eabcb1c6170abd3f03d23ce4c51 (diff) | |
download | ffmpeg-5d606863c353553488e88dfd89ea5571268d9dca.tar.gz |
lavc: remove disabled FF_API_OLD_DECODE_AUDIO cruft
-rw-r--r-- | libavcodec/avcodec.h | 65 | ||||
-rw-r--r-- | libavcodec/utils.c | 48 | ||||
-rw-r--r-- | libavcodec/version.h | 3 |
3 files changed, 0 insertions, 116 deletions
diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h index b3bdb61bbd..7bc7b618c7 100644 --- a/libavcodec/avcodec.h +++ b/libavcodec/avcodec.h @@ -487,11 +487,6 @@ typedef struct AVCodecDescriptor { */ #define AV_CODEC_PROP_LOSSLESS (1 << 2) -#if FF_API_OLD_DECODE_AUDIO -/* in bytes */ -#define AVCODEC_MAX_AUDIO_FRAME_SIZE 192000 // 1 second of 48khz 32bit audio -#endif - /** * @ingroup lavc_decoding * Required number of additionally allocated bytes at the end of the input bitstream for decoding. @@ -3360,66 +3355,6 @@ void avcodec_align_dimensions(AVCodecContext *s, int *width, int *height); void avcodec_align_dimensions2(AVCodecContext *s, int *width, int *height, int linesize_align[AV_NUM_DATA_POINTERS]); -#if FF_API_OLD_DECODE_AUDIO -/** - * Wrapper function which calls avcodec_decode_audio4. - * - * @deprecated Use avcodec_decode_audio4 instead. - * - * Decode the audio frame of size avpkt->size from avpkt->data into samples. - * Some decoders may support multiple frames in a single AVPacket, such - * decoders would then just decode the first frame. In this case, - * avcodec_decode_audio3 has to be called again with an AVPacket that contains - * the remaining data in order to decode the second frame etc. - * If no frame - * could be outputted, frame_size_ptr is zero. Otherwise, it is the - * decompressed frame size in bytes. - * - * @warning You must set frame_size_ptr to the allocated size of the - * output buffer before calling avcodec_decode_audio3(). - * - * @warning The input buffer must be FF_INPUT_BUFFER_PADDING_SIZE larger than - * the actual read bytes because some optimized bitstream readers read 32 or 64 - * bits at once and could read over the end. - * - * @warning The end of the input buffer avpkt->data should be set to 0 to ensure that - * no overreading happens for damaged MPEG streams. - * - * @warning You must not provide a custom get_buffer() when using - * avcodec_decode_audio3(). Doing so will override it with - * avcodec_default_get_buffer. Use avcodec_decode_audio4() instead, - * which does allow the application to provide a custom get_buffer(). - * - * @note You might have to align the input buffer avpkt->data and output buffer - * samples. The alignment requirements depend on the CPU: On some CPUs it isn't - * necessary at all, on others it won't work at all if not aligned and on others - * it will work but it will have an impact on performance. - * - * In practice, avpkt->data should have 4 byte alignment at minimum and - * samples should be 16 byte aligned unless the CPU doesn't need it - * (AltiVec and SSE do). - * - * @note Codecs which have the CODEC_CAP_DELAY capability set have a delay - * between input and output, these need to be fed with avpkt->data=NULL, - * avpkt->size=0 at the end to return the remaining frames. - * - * @param avctx the codec context - * @param[out] samples the output buffer, sample type in avctx->sample_fmt - * If the sample format is planar, each channel plane will - * be the same size, with no padding between channels. - * @param[in,out] frame_size_ptr the output buffer size in bytes - * @param[in] avpkt The input AVPacket containing the input buffer. - * You can create such packet with av_init_packet() and by then setting - * data and size, some decoders might in addition need other fields. - * All decoders are designed to use the least fields possible though. - * @return On error a negative value is returned, otherwise the number of bytes - * used or zero if no frame data was decompressed (used) from the input AVPacket. - */ -attribute_deprecated int avcodec_decode_audio3(AVCodecContext *avctx, int16_t *samples, - int *frame_size_ptr, - AVPacket *avpkt); -#endif - /** * Decode the audio frame of size avpkt->size from avpkt->data into frame. * diff --git a/libavcodec/utils.c b/libavcodec/utils.c index 25914282dc..ed39c0c3f1 100644 --- a/libavcodec/utils.c +++ b/libavcodec/utils.c @@ -1523,54 +1523,6 @@ int attribute_align_arg avcodec_decode_video2(AVCodecContext *avctx, AVFrame *pi return ret; } -#if FF_API_OLD_DECODE_AUDIO -int attribute_align_arg avcodec_decode_audio3(AVCodecContext *avctx, int16_t *samples, - int *frame_size_ptr, - AVPacket *avpkt) -{ - AVFrame frame = { { 0 } }; - int ret, got_frame = 0; - - if (avctx->get_buffer != avcodec_default_get_buffer) { - av_log(avctx, AV_LOG_ERROR, "Custom get_buffer() for use with" - "avcodec_decode_audio3() detected. Overriding with avcodec_default_get_buffer\n"); - av_log(avctx, AV_LOG_ERROR, "Please port your application to " - "avcodec_decode_audio4()\n"); - avctx->get_buffer = avcodec_default_get_buffer; - } - - ret = avcodec_decode_audio4(avctx, &frame, &got_frame, avpkt); - - if (ret >= 0 && got_frame) { - int ch, plane_size; - int planar = av_sample_fmt_is_planar(avctx->sample_fmt); - int data_size = av_samples_get_buffer_size(&plane_size, avctx->channels, - frame.nb_samples, - avctx->sample_fmt, 1); - if (*frame_size_ptr < data_size) { - av_log(avctx, AV_LOG_ERROR, "output buffer size is too small for " - "the current frame (%d < %d)\n", *frame_size_ptr, data_size); - return AVERROR(EINVAL); - } - - memcpy(samples, frame.extended_data[0], plane_size); - - if (planar && avctx->channels > 1) { - uint8_t *out = ((uint8_t *)samples) + plane_size; - for (ch = 1; ch < avctx->channels; ch++) { - memcpy(out, frame.extended_data[ch], plane_size); - out += plane_size; - } - } - *frame_size_ptr = data_size; - } else { - *frame_size_ptr = 0; - } - return ret; -} - -#endif - int attribute_align_arg avcodec_decode_audio4(AVCodecContext *avctx, AVFrame *frame, int *got_frame_ptr, diff --git a/libavcodec/version.h b/libavcodec/version.h index 76d686d6d4..f1cc74961a 100644 --- a/libavcodec/version.h +++ b/libavcodec/version.h @@ -49,9 +49,6 @@ #ifndef FF_API_REQUEST_CHANNELS #define FF_API_REQUEST_CHANNELS (LIBAVCODEC_VERSION_MAJOR < 56) #endif -#ifndef FF_API_OLD_DECODE_AUDIO -#define FF_API_OLD_DECODE_AUDIO (LIBAVCODEC_VERSION_MAJOR < 55) -#endif #ifndef FF_API_OLD_ENCODE_AUDIO #define FF_API_OLD_ENCODE_AUDIO (LIBAVCODEC_VERSION_MAJOR < 55) #endif |