ref: 5ae4049727d4f29036ad541bde5863c850aa7755
parent: 2991721da186f17df4930f77b940d66b8d27bc12
author: Ulrich Klauer <ulrich@chirlu.de>
date: Mon Feb 11 20:37:55 EST 2013
Remove deprecated ffmpeg format handler Remove the ffmpeg format handler that has been deprecated in commit 4568ed79319230843cf353faf79f973eb91b6e48 (refer to it for reasons).
--- a/CMakeLists.txt
+++ b/CMakeLists.txt
@@ -159,7 +159,6 @@
endif (NOT HAVE_SYS_SOUNDCARD_H)
optional(HAVE_WAVEAUDIO mmsystem.h winmm waveInGetDevCapsA waveaudio)
optional4(HAVE_OGG_VORBIS vorbis/codec.h ogg ogg_stream_flush vorbis vorbis_analysis_headerout vorbisfile ov_clear vorbisenc vorbis_encode_init_vbr vorbis)
-optional3(HAVE_FFMPEG ffmpeg/avformat.h avformat av_open_input_file avutil av_rescale_q avcodec avcodec_decode_audio2 ffmpeg)
optional(HAVE_WAVPACK wavpack/wavpack.h wavpack WavpackGetSampleRate wavpack)
if (HAVE_LAME_LAME_H OR HAVE_MAD_H)
--- a/ChangeLog
+++ b/ChangeLog
@@ -17,6 +17,7 @@
14.3.0 O -A/-U/-o/-i/-a/-g -e
14.4.0 E swap with parameters remix
14.4.1 OpenMP < 3.0 OpenMP >= 3.0
+ 14.4.1 F ffmpeg ffmpeg/avconv via pipe
Previously deprecated features (to be removed in future):
--- a/configure.ac
+++ b/configure.ac
@@ -541,10 +541,8 @@
fi
AC_OPTIONAL_FORMAT(sndfile, SNDFILE, [using_sndfile=$tmp_using_sndfile])
-AC_OPTIONAL_FORMAT(ffmpeg, FFMPEG, [SOX_PATH_FFMPEG(, using_ffmpeg=no)])
-
AC_OPTIONAL_FORMAT(oss, OSS, [AC_CHECK_HEADERS(sys/soundcard.h,,
[AC_CHECK_HEADERS(machine/soundcard.h,
[AC_CHECK_LIB(ossaudio, _oss_ioctl, OSS_LIBS="$OSS_LIBS -lossaudio")],
@@ -658,7 +656,6 @@
if test "x$using_amrwb" = "xyes"; then
echo " dlopen amrwb..............$enable_dl_amrwb"
fi
-echo "ffmpeg.....................$using_ffmpeg"
echo "flac.......................$using_flac"
echo "gsm........................$using_gsm $gsm_option"
echo "lpc10......................$using_lpc10 $lpc10_option"
--- a/soxformat.7
+++ b/soxformat.7
@@ -44,9 +44,7 @@
.SP
Format types that can be handled by an
external library via an optional pseudo file type (currently
-.B sndfile
-or
-.BR ffmpeg )
+.BR sndfile )
are marked e.g. `(also with \fB\-t sndfile\fR)'. This might be
useful if you have a file that doesn't work with SoX's default format
readers and writers, and there's an external reader or writer for that
@@ -315,14 +313,6 @@
See
.BR .paf .
.TP
-\fBffmpeg\fR (optional)
-This is a pseudo-type that forces ffmpeg to be used. The actual file
-type is deduced from the file name (it cannot be used on stdio).
-It can read a wide range of audio files, not all of which are
-documented here, and also the audio track of many video files
-(including AVI, WMV and MPEG). At present only the first audio track
-of a file can be read.
-.TP
\&\fB.flac\fR (optional; also with \fB\-t sndfile\fR)
Xiph.org's Free Lossless Audio CODEC compressed audio.
FLAC is an open, patent-free CODEC designed for compressing
@@ -464,12 +454,6 @@
See also
.B Ogg Vorbis
for a similar format.
-.TP
-\&\fB.mp4\fR, \fB.m4a\fR (optional)
-MP4 compressed audio. MP3 (MPEG 4) is part of the
-MPEG standards for audio and video compression. See
-.B mp3
-for more information.
.TP
\&\fB.nist\fR (also with \fB\-t sndfile\fR)
See \fB.sph\fR.
--- a/src/ffmpeg.c
+++ /dev/null
@@ -1,518 +1,0 @@
-/* libSoX ffmpeg formats.
- *
- * Copyright 2007, 2011 Reuben Thomas <rrt@sc3d.org>
- *
- * Based on ffplay.c and output_example.c Copyright 2003 Fabrice Bellard
- * Note: ffplay.c is distributed under the LGPL 2.1 or later;
- * output_example.c is under the MIT license:
- *-----------------------------------------------------------------------
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to deal
- * in the Software without restriction, including without limitation the rights
- * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
- * copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
- * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
- * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
- * THE SOFTWARE.
- *-----------------------------------------------------------------------
- *
- * This library is free software; you can redistribute it and/or modify it
- * under the terms of the GNU Lesser General Public License as published by
- * the Free Software Foundation; either version 2.1 of the License, or (at
- * your option) any later version.
- *
- * This library is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU Lesser
- * General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public License
- * along with this library; if not, write to the Free Software Foundation,
- * Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#include "sox_i.h"
-
-#ifdef HAVE_FFMPEG
-
-#include <assert.h>
-#include <stdio.h>
-#include <string.h>
-#include <ctype.h>
-#include "ffmpeg.h"
-
-#ifndef CODEC_TYPE_AUDIO
-#define CODEC_TYPE_AUDIO AVMEDIA_TYPE_AUDIO
-#endif
-#ifndef PKT_FLAG_KEY
-#define PKT_FLAG_KEY AV_PKT_FLAG_KEY
-#endif
-
-/* Private data for ffmpeg files */
-typedef struct {- int audio_index;
- int audio_stream;
- AVStream *audio_st;
- uint8_t *audio_buf_aligned;
- int audio_buf_index, audio_buf_size;
- int16_t *samples;
- int samples_index;
- AVOutputFormat *fmt;
- AVFormatContext *ctxt;
- int audio_input_frame_size;
- AVPacket audio_pkt;
- uint8_t *audio_buf_raw;
-} priv_t;
-
-/* open a given stream. Return 0 if OK */
-static int stream_component_open(priv_t * ffmpeg, int stream_index)
-{- AVFormatContext *ic = ffmpeg->ctxt;
- AVCodecContext *enc;
- AVCodec *codec;
-
- if (stream_index < 0 || stream_index >= (int)(ic->nb_streams))
- return -1;
- enc = ic->streams[stream_index]->codec;
-
- /* hack for AC3. XXX: suppress that */
- if (enc->channels > 2)
- enc->channels = 2;
-
- codec = avcodec_find_decoder(enc->codec_id);
- enc->workaround_bugs = 1;
-#if LIBAVCODEC_VERSION_INT < ((52<<16)+(0<<8)+0)
- enc->error_resilience = 1;
-#else
- enc->error_recognition = 1;
-#endif
-
- if (!codec || avcodec_open(enc, codec) < 0)
- return -1;
- if (enc->codec_type != AVMEDIA_TYPE_AUDIO) {- lsx_fail("ffmpeg CODEC %x is not an audio CODEC", enc->codec_type);- return -1;
- }
-
- ffmpeg->audio_stream = stream_index;
- ffmpeg->audio_st = ic->streams[stream_index];
- ffmpeg->audio_buf_size = 0;
- ffmpeg->audio_buf_index = 0;
-
- memset(&ffmpeg->audio_pkt, 0, sizeof(ffmpeg->audio_pkt));
-
- return 0;
-}
-
-static void stream_component_close(priv_t * ffmpeg, int stream_index)
-{- AVFormatContext *ic = ffmpeg->ctxt;
- AVCodecContext *enc;
-
- if (stream_index < 0 || stream_index >= (int)ic->nb_streams)
- return;
- enc = ic->streams[stream_index]->codec;
-
- avcodec_close(enc);
-}
-
-/* Decode one audio frame and returns its uncompressed size */
-static int audio_decode_frame(priv_t * ffmpeg, uint8_t *audio_buf, int buf_size)
-{- AVPacket *pkt = &ffmpeg->audio_pkt;
- int len1, data_size;
-
- for (;;) {- /* NOTE: the audio packet can contain several frames */
- while (ffmpeg->audio_pkt.size > 0) {- data_size = buf_size;
- len1 = avcodec_decode_audio3(ffmpeg->audio_st->codec,
- (int16_t *)audio_buf, &data_size,
- pkt);
- if (len1 < 0) /* if error, we skip the rest of the packet */
- return 0;
-
- ffmpeg->audio_pkt.data += len1;
- ffmpeg->audio_pkt.size -= len1;
- if (data_size <= 0)
- continue;
- return data_size;
- }
- }
-}
-
-/* On some platforms, libavcodec wants the output buffer aligned to 16
- * bytes (because it uses SSE/Altivec internally). */
-#define ALIGN16(p) ((uint8_t *)(p) + (16 - (size_t)(p) % 16))
-
-static int startread(sox_format_t * ft)
-{- priv_t * ffmpeg = (priv_t *)ft->priv;
- AVFormatParameters params;
- int ret;
- int i;
-
- ffmpeg->audio_buf_raw = lsx_calloc(1, (size_t)AVCODEC_MAX_AUDIO_FRAME_SIZE + 32);
- ffmpeg->audio_buf_aligned = ALIGN16(ffmpeg->audio_buf_raw);
-
- /* Signal audio stream not found */
- ffmpeg->audio_index = -1;
-
- /* register all CODECs, demux and protocols */
- av_register_all();
-
- /* Open file and get format */
- memset(¶ms, 0, sizeof(params));
- if ((ret = av_open_input_file(&ffmpeg->ctxt, ft->filename, NULL, 0, ¶ms)) < 0) {- lsx_fail("ffmpeg cannot open file for reading: %s (code %d)", ft->filename, ret);- return SOX_EOF;
- }
-
- /* Get CODEC parameters */
- if ((ret = av_find_stream_info(ffmpeg->ctxt)) < 0) {- lsx_fail("ffmpeg could not find CODEC parameters for %s", ft->filename);- return SOX_EOF;
- }
-
- /* Now we can begin to play (RTSP stream only) */
- av_read_play(ffmpeg->ctxt);
-
- /* Find audio stream (FIXME: allow different stream to be selected) */
- for (i = 0; (unsigned)i < ffmpeg->ctxt->nb_streams; i++) {- AVCodecContext *enc = ffmpeg->ctxt->streams[i]->codec;
- if (enc->codec_type == AVMEDIA_TYPE_AUDIO && ffmpeg->audio_index < 0) {- ffmpeg->audio_index = i;
- break;
- }
- }
-
- /* Open the stream */
- if (ffmpeg->audio_index < 0 ||
- stream_component_open(ffmpeg, ffmpeg->audio_index) < 0 ||
- ffmpeg->audio_stream < 0) {- lsx_fail("ffmpeg could not open CODECs for %s", ft->filename);- return SOX_EOF;
- }
-
- /* Copy format info */
- ft->signal.rate = ffmpeg->audio_st->codec->sample_rate;
- ft->encoding.bits_per_sample = 16;
- ft->encoding.encoding = SOX_ENCODING_SIGN2;
- ft->signal.channels = ffmpeg->audio_st->codec->channels;
- ft->signal.length = 0; /* Currently we can't seek; no idea how to get this
- info from ffmpeg anyway (in time, yes, but not in
- samples); but ffmpeg *can* seek */
-
- return SOX_SUCCESS;
-}
-
-/*
- * Read up to len samples of type sox_sample_t from file into buf[].
- * Return number of samples read.
- */
-static size_t read_samples(sox_format_t * ft, sox_sample_t *buf, size_t len)
-{- priv_t * ffmpeg = (priv_t *)ft->priv;
- AVPacket *pkt = &ffmpeg->audio_pkt;
- int ret;
- size_t nsamp = 0, nextra;
-
- /* Read data repeatedly until buf is full or no more can be read */
- do {- /* If input buffer empty, read more data */
- if (ffmpeg->audio_buf_index * 2 >= ffmpeg->audio_buf_size) {- if ((ret = av_read_frame(ffmpeg->ctxt, pkt)) < 0 &&
- (ret == AVERROR_EOF || url_ferror(ffmpeg->ctxt->pb)))
- break;
- ffmpeg->audio_buf_size = audio_decode_frame(ffmpeg, ffmpeg->audio_buf_aligned, AVCODEC_MAX_AUDIO_FRAME_SIZE);
- ffmpeg->audio_buf_index = 0;
- }
-
- /* Convert data into SoX samples up to size of buffer */
- nextra = min((ffmpeg->audio_buf_size - ffmpeg->audio_buf_index) / 2, (int)(len - nsamp));
- for (; nextra > 0; nextra--)
- buf[nsamp++] = SOX_SIGNED_16BIT_TO_SAMPLE(((int16_t *)ffmpeg->audio_buf_aligned)[ffmpeg->audio_buf_index++], ft->clips);
- } while (nsamp < len && nextra > 0);
-
- return nsamp;
-}
-
-/*
- * Close file for ffmpeg (this doesn't close the file handle)
- */
-static int stopread(sox_format_t * ft)
-{- priv_t * ffmpeg = (priv_t *)ft->priv;
-
- if (ffmpeg->audio_stream >= 0)
- stream_component_close(ffmpeg, ffmpeg->audio_stream);
- if (ffmpeg->ctxt) {- av_close_input_file(ffmpeg->ctxt);
- ffmpeg->ctxt = NULL; /* safety */
- }
-
- free(ffmpeg->audio_buf_raw);
- return SOX_SUCCESS;
-}
-
-/*
- * add an audio output stream
- */
-static AVStream *add_audio_stream(sox_format_t * ft, AVFormatContext *oc, enum CodecID codec_id)
-{- AVCodecContext *c;
- AVStream *st;
-
- st = av_new_stream(oc, 1);
- if (!st) {- lsx_fail("ffmpeg could not alloc stream");- return NULL;
- }
-
- c = st->codec;
- c->codec_id = codec_id;
- c->codec_type = AVMEDIA_TYPE_AUDIO;
-
- /* put sample parameters */
- c->bit_rate = 256000; /* FIXME: allow specification */
- /* FIXME: currently mplayer says files do not have a specified
- compressed bit-rate */
- c->sample_rate = ft->signal.rate;
- c->channels = ft->signal.channels;
- return st;
-}
-
-static int open_audio(priv_t * ffmpeg, AVStream *st)
-{- AVCodecContext *c;
- AVCodec *codec;
-
- c = st->codec;
-
- /* find the audio encoder */
- codec = avcodec_find_encoder(c->codec_id);
- if (!codec) {- lsx_fail("ffmpeg CODEC not found");- return SOX_EOF;
- }
-
- /* open it */
- if (avcodec_open(c, codec) < 0) {- lsx_fail("ffmpeg could not open CODEC");- return SOX_EOF;
- }
-
- ffmpeg->audio_buf_raw = lsx_malloc((size_t)AVCODEC_MAX_AUDIO_FRAME_SIZE + 32);
- ffmpeg->audio_buf_aligned = ALIGN16(ffmpeg->audio_buf_raw);
-
- /* ugly hack for PCM codecs (will be removed ASAP with new PCM
- support to compute the input frame size in samples */
- if (c->frame_size <= 1) {- ffmpeg->audio_input_frame_size = AVCODEC_MAX_AUDIO_FRAME_SIZE / c->channels;
- switch(st->codec->codec_id) {- case CODEC_ID_PCM_S16LE:
- case CODEC_ID_PCM_S16BE:
- case CODEC_ID_PCM_U16LE:
- case CODEC_ID_PCM_U16BE:
- ffmpeg->audio_input_frame_size >>= 1;
- break;
- default:
- break;
- }
- } else
- ffmpeg->audio_input_frame_size = c->frame_size;
-
- ffmpeg->samples = lsx_malloc((size_t)(ffmpeg->audio_input_frame_size * 2 * c->channels));
-
- return SOX_SUCCESS;
-}
-
-static int startwrite(sox_format_t * ft)
-{- priv_t * ffmpeg = (priv_t *)ft->priv;
-
- /* initialize libavcodec, and register all codecs and formats */
- av_register_all();
-
- /* auto detect the output format from the name. default is
- mpeg. */
- ffmpeg->fmt = av_guess_format(NULL, ft->filename, NULL);
- if (!ffmpeg->fmt) {- lsx_warn("ffmpeg could not deduce output format from file extension; using MPEG");- ffmpeg->fmt = av_guess_format("mpeg", NULL, NULL);- if (!ffmpeg->fmt) {- lsx_fail("ffmpeg could not find suitable output format");- return SOX_EOF;
- }
- }
-
- /* allocate the output media context */
- ffmpeg->ctxt = avformat_alloc_context();
- if (!ffmpeg->ctxt) {- fprintf(stderr, "ffmpeg out of memory error");
- return SOX_EOF;
- }
- ffmpeg->ctxt->oformat = ffmpeg->fmt;
- snprintf(ffmpeg->ctxt->filename, sizeof(ffmpeg->ctxt->filename), "%s", ft->filename);
-
- /* add the audio stream using the default format codecs
- and initialize the codecs */
- ffmpeg->audio_st = NULL;
- if (ffmpeg->fmt->audio_codec != CODEC_ID_NONE) {- ffmpeg->audio_st = add_audio_stream(ft, ffmpeg->ctxt, ffmpeg->fmt->audio_codec);
- if (ffmpeg->audio_st == NULL)
- return SOX_EOF;
- }
-
- /* set the output parameters (must be done even if no
- parameters). */
- if (av_set_parameters(ffmpeg->ctxt, NULL) < 0) {- lsx_fail("ffmpeg invalid output format parameters");- return SOX_EOF;
- }
-
- /* Next line for debugging */
- /* dump_format(ffmpeg->ctxt, 0, ft->filename, 1); */
-
- /* now that all the parameters are set, we can open the audio and
- codec and allocate the necessary encode buffers */
- if (ffmpeg->audio_st)
- if (open_audio(ffmpeg, ffmpeg->audio_st) == SOX_EOF)
- return SOX_EOF;
-
- /* open the output file, if needed */
- if (!(ffmpeg->fmt->flags & AVFMT_NOFILE)) {- if (url_fopen(&ffmpeg->ctxt->pb, ft->filename, URL_WRONLY) < 0) {- lsx_fail("ffmpeg could not open `%s'", ft->filename);- return SOX_EOF;
- }
- }
-
- /* write the stream header, if any */
- av_write_header(ffmpeg->ctxt);
-
- return SOX_SUCCESS;
-}
-
-/*
- * Write up to len samples of type sox_sample_t from buf[] into file.
- * Return number of samples written.
- */
-static size_t write_samples(sox_format_t * ft, const sox_sample_t *buf, size_t len)
-{- priv_t * ffmpeg = (priv_t *)ft->priv;
- size_t nread = 0, nwritten = 0;
-
- /* Write data repeatedly until buf is empty */
- do {- /* If output frame is not full, copy data into it */
- if (ffmpeg->samples_index < ffmpeg->audio_input_frame_size) {- SOX_SAMPLE_LOCALS;
- for (; nread < len && ffmpeg->samples_index < ffmpeg->audio_input_frame_size; nread++)
- ffmpeg->samples[ffmpeg->samples_index++] = SOX_SAMPLE_TO_SIGNED_16BIT(buf[nread], ft->clips);
- }
-
- /* If output frame full or no more data to read, write it out */
- if (ffmpeg->samples_index == ffmpeg->audio_input_frame_size ||
- (len == 0 && ffmpeg->samples_index > 0)) {- AVCodecContext *c = ffmpeg->audio_st->codec;
- AVPacket pkt;
-
- av_init_packet(&pkt);
- pkt.size = avcodec_encode_audio(c, ffmpeg->audio_buf_aligned, AVCODEC_MAX_AUDIO_FRAME_SIZE, ffmpeg->samples);
- pkt.pts = av_rescale_q(c->coded_frame->pts, c->time_base, ffmpeg->audio_st->time_base);
- pkt.flags |= AV_PKT_FLAG_KEY;
- pkt.stream_index = ffmpeg->audio_st->index;
- pkt.data = ffmpeg->audio_buf_aligned;
-
- /* write the compressed frame to the media file */
- if (av_write_frame(ffmpeg->ctxt, &pkt) != 0)
- lsx_fail("ffmpeg had error while writing audio frame");-
- /* Increment nwritten whether write succeeded or not; we have to
- get rid of the input! */
- nwritten += ffmpeg->samples_index;
- ffmpeg->samples_index = 0;
- }
- } while (nread < len);
-
- return nwritten;
-}
-
-/*
- * Close file for ffmpeg (this doesn't close the file handle)
- */
-static int stopwrite(sox_format_t * ft)
-{- priv_t * ffmpeg = (priv_t *)ft->priv;
- int i;
-
- /* Close CODEC */
- if (ffmpeg->audio_st) {- avcodec_close(ffmpeg->audio_st->codec);
- }
-
- free(ffmpeg->samples);
- free(ffmpeg->audio_buf_raw);
-
- /* Write the trailer, if any */
- av_write_trailer(ffmpeg->ctxt);
-
- /* Free the streams */
- for (i = 0; (unsigned)i < ffmpeg->ctxt->nb_streams; i++) {- av_freep(&ffmpeg->ctxt->streams[i]->codec);
- av_freep(&ffmpeg->ctxt->streams[i]);
- }
-
- if (!(ffmpeg->fmt->flags & AVFMT_NOFILE)) {- /* close the output file */
-#if (LIBAVFORMAT_VERSION_INT < 0x340000)
- url_fclose(&ffmpeg->ctxt->pb);
-#else
- url_fclose(ffmpeg->ctxt->pb);
-#endif
- }
-
- /* Free the output context */
- av_free(ffmpeg->ctxt);
-
- return SOX_SUCCESS;
-}
-
-LSX_FORMAT_HANDLER(ffmpeg)
-{- /* Format file suffixes */
- /* For now, comment out formats built in to SoX */
- static char const * const names[] = {- "ffmpeg", /* special type to force use of ffmpeg */
- "mp4",
- "m4a",
- "m4b",
- "avi",
- "wmv",
- "mpg",
- NULL
- };
-
- static unsigned const write_encodings[] = {SOX_ENCODING_SIGN2, 16, 0, 0};-
- static sox_format_handler_t handler = {SOX_LIB_VERSION_CODE,- "Pseudo format to use libffmpeg", names, SOX_FILE_NOSTDIO,
- startread, read_samples, stopread,
- startwrite, write_samples, stopwrite,
- NULL, write_encodings, NULL, sizeof(priv_t)
- };
-
- return &handler;
-}
-
-#endif /* HAVE_FFMPEG */
--- a/src/ffmpeg.h
+++ /dev/null
@@ -1,34 +1,0 @@
-/* This library is free software; you can redistribute it and/or modify it
- * under the terms of the GNU Lesser General Public License as published by
- * the Free Software Foundation; either version 2.1 of the License, or (at
- * your option) any later version.
- *
- * This library is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU Lesser
- * General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public License
- * along with this library; if not, write to the Free Software Foundation,
- * Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#if defined __GNUC__
- #pragma GCC system_header
-#elif defined __SUNPRO_C
- #pragma disable_warn
-#elif defined _MSC_VER
- #pragma warning(push, 1)
-#endif
-
-#if HAVE_LIBAVFORMAT_AVFORMAT_H
-#include <libavformat/avformat.h>
-#else
-#include <ffmpeg/avformat.h>
-#endif
-
-#if defined __SUNPRO_C
- #pragma enable_warn
-#elif defined _MSC_VER
- #pragma warning(pop)
-#endif
--- a/src/formats.h
+++ b/src/formats.h
@@ -80,9 +80,6 @@
#if defined HAVE_COREAUDIO && (defined STATIC_COREAUDIO || !defined HAVE_LIBLTDL)
FORMAT(coreaudio)
#endif
-#if defined HAVE_FFMPEG && (defined STATIC_FFMPEG || !defined HAVE_LIBLTDL)
- FORMAT(ffmpeg)
-#endif
#if defined HAVE_FLAC && (defined STATIC_FLAC || !defined HAVE_LIBLTDL)
FORMAT(flac)
#endif
--- a/src/optional-fmts.am
+++ b/src/optional-fmts.am
@@ -75,21 +75,6 @@
endif
endif
-if HAVE_FFMPEG
-if STATIC_FFMPEG
- libsox_la_SOURCES += ffmpeg.c ffmpeg.h
-if STATIC_LIBSOX_ONLY
- sox_LDADD += @FFMPEG_LIBS@
-else
- libsox_la_LIBADD += @FFMPEG_LIBS@
-endif
-else
- libsox_fmt_ffmpeg_la_SOURCES = ffmpeg.c ffmpeg.h
- libsox_fmt_ffmpeg_la_LIBADD = libsox.la @FFMPEG_LIBS@
- pkglib_LTLIBRARIES += libsox_fmt_ffmpeg.la
-endif
-endif
-
if HAVE_FLAC
if STATIC_FLAC
libsox_la_SOURCES += flac.c
--- a/src/soxconfig.h.cmake
+++ b/src/soxconfig.h.cmake
@@ -9,7 +9,6 @@
#cmakedefine HAVE_BYTESWAP_H 1
#cmakedefine HAVE_COREAUDIO 1
#cmakedefine HAVE_FENV_H 1
-#cmakedefine HAVE_FFMPEG 1
#cmakedefine HAVE_FLAC 1
#cmakedefine HAVE_FMEMOPEN 1
#cmakedefine HAVE_FSEEKO 1