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@@ -0,0 +1,322 @@
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Index: a/decoder_plugins/ffmpeg/ffmpeg.c
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===================================================================
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diff --git a/configure.in b/configure.in
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index c0243be..b781a72 100644
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--- a/configure.in
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+++ b/configure.in
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@@ -8,6 +8,7 @@ AC_CONFIG_HEADERS([config.h])
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AM_INIT_AUTOMAKE
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AC_PREREQ(2.60)
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+AC_USE_SYSTEM_EXTENSIONS
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m4_version_prereq([2.64], , [AUTOCONF_DEPRECATED='yes'])
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diff --git a/decoder_plugins/ffmpeg/ffmpeg.c b/decoder_plugins/ffmpeg/ffmpeg.c
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index 7b90493..8452d26 100644
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--- a/decoder_plugins/ffmpeg/ffmpeg.c
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+++ b/decoder_plugins/ffmpeg/ffmpeg.c
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@@ -26,6 +26,7 @@
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#include <stdlib.h>
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#include <pthread.h>
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#include <string.h>
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+#include <strings.h>
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#include <assert.h>
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#include <stdint.h>
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@@ -81,6 +82,7 @@ GCC_DIAG_ON(deprecated-declarations)
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# include <ffmpeg/avformat.h>
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#endif
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+#include <libavcodec/avcodec.h>
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/* FFmpeg also likes common names, without that, our common.h and log.h
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* would not be included. */
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@@ -199,6 +201,7 @@ struct ffmpeg_data
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AVStream *stream;
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AVCodecContext *enc;
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AVCodec *codec;
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+ AVFrame *frame;
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#ifndef HAVE_AVCODEC_DECODE_AUDIO4
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int avbuf_size;
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@@ -226,6 +229,8 @@ struct ffmpeg_data
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#if SEEK_IN_DECODER && defined(DEBUG)
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pthread_t thread_id;
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#endif
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+
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+ int64_t cur_dts; /* current stream timestamp */
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};
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struct extn_list {
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@@ -312,7 +317,7 @@ static unsigned int find_first_audio (AVFormatContext *ic)
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#if LIBAVUTIL_VERSION_INT < AV_VERSION_INT(50,15,1)
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if (ic->streams[result]->codec->codec_type == CODEC_TYPE_AUDIO)
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#else
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- if (ic->streams[result]->codec->codec_type == AVMEDIA_TYPE_AUDIO)
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+ if (ic->streams[result]->codecpar->codec_type == AVMEDIA_TYPE_AUDIO)
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#endif
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{
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break;
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@@ -510,8 +515,6 @@ static void ffmpeg_init ()
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av_log_set_level (AV_LOG_ERROR);
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#endif
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av_log_set_callback (ffmpeg_log_cb);
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- avcodec_register_all ();
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- av_register_all ();
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supported_extns = lists_strs_new (16);
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load_audio_extns (supported_extns);
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@@ -861,19 +864,10 @@ GCC_DIAG_ON(deprecated-declarations)
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/* Downmix multi-channel audios to stereo. */
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static void set_downmixing (struct ffmpeg_data *data)
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{
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-#ifdef HAVE_AV_GET_CHANNEL_LAYOUT_NB_CHANNELS
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- if (av_get_channel_layout_nb_channels (data->enc->channel_layout) <= 2)
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- return;
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-#else
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- if (data->enc->channels <= 2)
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+ if (data->enc->ch_layout.nb_channels <= 2)
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return;
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-#endif
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-#ifdef HAVE_STRUCT_AVCODECCONTEXT_REQUEST_CHANNELS
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- set_request_channels (data->enc, 2);
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-#else
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- data->enc->request_channel_layout = AV_CH_LAYOUT_STEREO;
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-#endif
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+ av_channel_layout_copy(&data->enc->ch_layout, &(AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO);
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}
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static void *ffmpeg_open (const char *file)
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@@ -889,24 +883,11 @@ static void *ffmpeg_open (const char *file)
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data->stream = NULL;
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data->enc = NULL;
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data->codec = NULL;
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+ data->frame = NULL;
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data->sample_width = 0;
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data->bitrate = 0;
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data->avg_bitrate = 0;
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-#if !defined(HAVE_AVCODEC_DECODE_AUDIO4) && defined(HAVE_POSIX_MEMALIGN)
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- /* The sample buffer should be 16 byte aligned (because of SSE); a
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- * segmentation fault may occur otherwise. We allocate it dynamically
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- * (to avoid overflowing OpenBSD's default pthread stack size) and once
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- * at open (to avoid delays on each decode call).
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- *
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- * See: avcodec.h in ffmpeg
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- */
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- data->avbuf_size = AVCODEC_MAX_AUDIO_FRAME_SIZE * 3 / 2;
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- err = posix_memalign ((void **)&data->avbuf, 16, data->avbuf_size);
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- if (err)
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- fatal ("Can't allocate memory!");
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-#endif
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-
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data->remain_buf = NULL;
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data->remain_buf_len = 0;
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data->delay = false;
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@@ -965,9 +946,8 @@ static void *ffmpeg_open (const char *file)
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}
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data->stream = data->ic->streams[audio_ix];
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- data->enc = data->stream->codec;
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- data->codec = avcodec_find_decoder (data->enc->codec_id);
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+ data->codec = avcodec_find_decoder (data->stream->codecpar->codec_id);
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if (!data->codec) {
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decoder_error (&data->error, ERROR_FATAL, 0,
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"No codec for this file");
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@@ -984,7 +964,7 @@ static void *ffmpeg_open (const char *file)
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* FFmpeg/LibAV in use. For some versions this will be caught in
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* *_find_stream_info() above and misreported as an unfound codec
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* parameters error. */
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- if (data->codec->capabilities & CODEC_CAP_EXPERIMENTAL) {
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+ if (data->codec->capabilities & AV_CODEC_CAP_EXPERIMENTAL) {
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decoder_error (&data->error, ERROR_FATAL, 0,
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"The codec is experimental and may damage MOC: %s",
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data->codec->name);
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@@ -992,21 +972,23 @@ static void *ffmpeg_open (const char *file)
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}
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#endif
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- set_downmixing (data);
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- if (data->codec->capabilities & CODEC_CAP_TRUNCATED)
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- data->enc->flags |= CODEC_FLAG_TRUNCATED;
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+ data->enc = avcodec_alloc_context3(data->codec);
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+ if (!data->enc) {
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+ decoder_error (&data->error, ERROR_FATAL, 0,
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+ "Could not allocate audio context");
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+ goto end;
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+ }
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+ avcodec_parameters_to_context(data->enc, data->stream->codecpar);
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-#ifdef HAVE_AVCODEC_OPEN2
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if (avcodec_open2 (data->enc, data->codec, NULL) < 0)
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-#else
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- if (avcodec_open (data->enc, data->codec) < 0)
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-#endif
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{
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decoder_error (&data->error, ERROR_FATAL, 0,
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"No codec for this file");
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goto end;
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}
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+ set_downmixing (data);
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+
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data->fmt = fmt_from_codec (data);
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if (data->fmt == 0)
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data->fmt = fmt_from_sample_fmt (data);
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@@ -1025,7 +1007,7 @@ static void *ffmpeg_open (const char *file)
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data->sample_width = sfmt_Bps (data->fmt);
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- if (data->codec->capabilities & CODEC_CAP_DELAY)
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+ if (data->codec->capabilities & AV_CODEC_CAP_DELAY)
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data->delay = true;
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data->seek_broken = is_seek_broken (data);
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data->timing_broken = is_timing_broken (data->ic);
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@@ -1148,7 +1130,6 @@ static inline AVPacket *new_packet (struct ffmpeg_data *data)
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assert (data->stream);
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pkt = (AVPacket *)xmalloc (sizeof (AVPacket));
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- av_init_packet (pkt);
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pkt->data = NULL;
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pkt->size = 0;
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pkt->stream_index = data->stream->index;
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@@ -1210,68 +1191,54 @@ static AVPacket *get_packet (struct ffmpeg_data *data)
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return NULL;
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}
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-#ifndef HAVE_AVCODEC_DECODE_AUDIO4
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-/* Decode samples from packet data using pre-avcodec_decode_audio4(). */
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+/* Decode samples from packet data using new ffmpeg API. */
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static int decode_packet (struct ffmpeg_data *data, AVPacket *pkt,
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char *buf, int buf_len)
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{
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- int filled = 0, len, data_size, copied;
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+ int filled = 0, i, ch, data_size, copied, ret;
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-#ifndef HAVE_POSIX_MEMALIGN
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- /* The sample buffer should be 16 byte aligned (because SSE), a segmentation
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- * fault may occur otherwise.
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- *
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- * See: avcodec.h in ffmpeg
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- */
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- char avbuf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2] __attribute__((aligned(16)));
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+ if (!data->frame) {
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+ if (!(data->frame = av_frame_alloc())) {
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+ decoder_error (&data->error, ERROR_STREAM, 0,
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+ "Error submitting packet to the decoder");
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+ return filled;
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+ }
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+ }
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- data->avbuf_size = sizeof(avbuf);
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- data->avbuf = avbuf;
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-#endif
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+ ret = avcodec_send_packet(data->enc, pkt);
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+ if (ret < 0) {
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+ decoder_error (&data->error, ERROR_STREAM, 0,
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+ "Error submitting packet to the decoder");
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+ return filled;
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+ }
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- do {
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- data_size = data->avbuf_size;
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-
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-#if defined(HAVE_AVCODEC_DECODE_AUDIO3)
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- len = avcodec_decode_audio3 (data->enc, (int16_t *)data->avbuf,
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- &data_size, pkt);
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-#elif defined(HAVE_AVCODEC_DECODE_AUDIO2)
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- len = avcodec_decode_audio2 (data->enc, (int16_t *)data->avbuf,
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- &data_size, pkt->data, pkt->size);
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-#else
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- len = avcodec_decode_audio (data->enc, (int16_t *)data->avbuf,
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- &data_size, pkt->data, pkt->size);
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-#endif
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+ data_size = av_get_bytes_per_sample(data->enc->sample_fmt);
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- if (len < 0) {
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- /* skip frame */
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+ while (ret >= 0) {
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+ ret = avcodec_receive_frame(data->enc, data->frame);
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+ if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
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+ break;
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+ } else if (ret < 0) {
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decoder_error (&data->error, ERROR_STREAM, 0,
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"Error in the stream!");
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break;
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}
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- debug ("Decoded %dB", data_size);
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-
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- pkt->data += len;
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- pkt->size -= len;
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+ for (i = 0; i < data->frame->nb_samples; i++) {
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+ for (ch = 0; ch < data->enc->channels; ch++) {
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+ copied = copy_or_buffer (data,
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+ data->frame->data[ch] + data_size*i, data_size,
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+ buf, buf_len);
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- if (data->eof && data_size == 0) {
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- data->eos = true;
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- break;
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+ buf += copied;
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+ filled += copied;
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+ buf_len -= copied;
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+ }
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}
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-
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- copied = copy_or_buffer (data, data->avbuf, data_size, buf, buf_len);
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-
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- buf += copied;
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- filled += copied;
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- buf_len -= copied;
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-
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- debug ("Copying %dB (%dB filled)", copied, filled);
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- } while (pkt->size > 0);
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+ }
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return filled;
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}
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-#endif
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#ifdef HAVE_AVCODEC_DECODE_AUDIO4
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/* Decode samples from packet data using avcodec_decode_audio4(). */
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@@ -1393,7 +1360,7 @@ static bool seek_in_stream (struct ffmpeg_data *data, int sec)
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seek_ts += data->stream->start_time;
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}
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- if (data->stream->cur_dts > seek_ts)
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+ if (data->cur_dts > seek_ts)
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flags |= AVSEEK_FLAG_BACKWARD;
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rc = av_seek_frame (data->ic, data->stream->index, seek_ts, flags);
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@@ -1403,7 +1370,7 @@ static bool seek_in_stream (struct ffmpeg_data *data, int sec)
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return false;
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}
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- avcodec_flush_buffers (data->stream->codec);
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+ avcodec_flush_buffers (data->enc);
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return true;
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}
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@@ -1471,6 +1438,7 @@ static int ffmpeg_decode (void *prv_data, char *buf, int buf_len,
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continue;
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}
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#endif
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+ data->cur_dts = pkt->dts;
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saved_pkt_data_ptr = pkt->data;
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bytes_used += pkt->size;
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@@ -1535,10 +1503,6 @@ static void ffmpeg_close (void *prv_data)
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free_remain_buf (data);
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}
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-#if !defined(HAVE_AVCODEC_DECODE_AUDIO4) && defined(HAVE_POSIX_MEMALIGN)
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- free (data->avbuf);
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-#endif
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-
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ffmpeg_log_repeats (NULL);
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decoder_error_clear (&data->error);
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free (data);
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Luis