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authorMichael Niedermayer <michaelni@gmx.at>2012-04-20 22:18:26 +0200
committerMichael Niedermayer <michaelni@gmx.at>2012-04-20 22:18:26 +0200
commit3194ab78a6c4ea0a4c60c91c4d0ea34028ca408f (patch)
tree13f41910a7e4feec8d64182bf3b6772d88236098
parent9b1f776d751472e8a376b412d02a96a35044e2a0 (diff)
parentb0e9edc44f1722787adacbff9aa60343206a58c0 (diff)
Merge remote-tracking branch 'qatar/master'
* qatar/master: avcodec: add a cook parser to get subpacket duration FATE: allow lavf tests to alter input parameters FATE: replace the acodec-pcm_s24daud test with an enc_dec_pcm checksum test FATE: replace the acodec-g726 test with 4 new encode/decode tests FATE: replace current g722 encoding tests with an encode/decode test FATE: add a pattern rule for generating asynth wav files FATE: optionally write a WAVE header in audiogen avutil: add audio fifo buffer Conflicts: doc/APIchanges libavcodec/version.h libavutil/avutil.h tests/Makefile tests/codec-regression.sh tests/fate/voice.mak tests/lavf-regression.sh tests/ref/acodec/g722 tests/ref/acodec/g726 tests/ref/acodec/pcm_s24daud tests/ref/lavf/dv_fmt tests/ref/lavf/gxf tests/ref/lavf/mxf tests/ref/lavf/mxf_d10 tests/ref/seek/lavf_dv Merged-by: Michael Niedermayer <michaelni@gmx.at>
-rw-r--r--doc/APIchanges12
-rw-r--r--libavcodec/Makefile1
-rw-r--r--libavcodec/allcodecs.c1
-rw-r--r--libavcodec/cook_parser.c59
-rw-r--r--libavcodec/version.h4
-rw-r--r--libavformat/daud.c2
-rw-r--r--libavformat/rmdec.c1
-rw-r--r--libavutil/Makefile2
-rw-r--r--libavutil/audio_fifo.c193
-rw-r--r--libavutil/audio_fifo.h146
-rw-r--r--libavutil/avutil.h2
-rw-r--r--tests/Makefile5
-rw-r--r--tests/audiogen.c56
-rwxr-xr-xtests/codec-regression.sh13
-rw-r--r--tests/fate/pcm.mak5
-rw-r--r--tests/fate/voice.mak35
-rwxr-xr-xtests/lavf-regression.sh34
-rw-r--r--tests/ref/acodec/g7224
-rw-r--r--tests/ref/acodec/g7264
-rw-r--r--tests/ref/acodec/pcm_s24daud4
-rw-r--r--tests/ref/fate/dcinema-encode1
-rw-r--r--tests/ref/fate/g722-encode1
-rw-r--r--tests/ref/fate/g722enc1
-rw-r--r--tests/ref/fate/g726-encode-2bit1
-rw-r--r--tests/ref/fate/g726-encode-3bit1
-rw-r--r--tests/ref/fate/g726-encode-4bit1
-rw-r--r--tests/ref/fate/g726-encode-5bit1
-rw-r--r--tests/ref/lavf/dv_fmt4
-rw-r--r--tests/ref/lavf/gxf4
-rw-r--r--tests/ref/lavf/mxf_d104
-rw-r--r--tests/ref/seek/g726_wav53
-rw-r--r--tests/ref/seek/pcm_s24daud_30227
32 files changed, 536 insertions, 146 deletions
diff --git a/doc/APIchanges b/doc/APIchanges
index ab78401488..a0b786d79e 100644
--- a/doc/APIchanges
+++ b/doc/APIchanges
@@ -22,6 +22,18 @@ API changes, most recent first:
2012-03-26 - a67d9cf - lavfi 2.66.100
Add avfilter_fill_frame_from_{audio_,}buffer_ref() functions.
+2012-xx-xx - xxxxxxx - lavu 51.28.0 - audio_fifo.h
+ Add audio FIFO functions:
+ av_audio_fifo_free()
+ av_audio_fifo_alloc()
+ av_audio_fifo_realloc()
+ av_audio_fifo_write()
+ av_audio_fifo_read()
+ av_audio_fifo_drain()
+ av_audio_fifo_reset()
+ av_audio_fifo_size()
+ av_audio_fifo_space()
+
2012-xx-xx - xxxxxxx - lavfi 2.16.0 - avfiltergraph.h
Add avfilter_graph_parse2()
diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index d7179c2233..e336917231 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -679,6 +679,7 @@ OBJS-$(CONFIG_AC3_PARSER) += ac3_parser.o ac3tab.o \
aac_ac3_parser.o
OBJS-$(CONFIG_ADX_PARSER) += adx_parser.o adx.o
OBJS-$(CONFIG_CAVSVIDEO_PARSER) += cavs_parser.o
+OBJS-$(CONFIG_COOK_PARSER) += cook_parser.o
OBJS-$(CONFIG_DCA_PARSER) += dca_parser.o
OBJS-$(CONFIG_DIRAC_PARSER) += dirac_parser.o
OBJS-$(CONFIG_DNXHD_PARSER) += dnxhd_parser.o
diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c
index 4a24d3199d..c8a6621818 100644
--- a/libavcodec/allcodecs.c
+++ b/libavcodec/allcodecs.c
@@ -437,6 +437,7 @@ void avcodec_register_all(void)
REGISTER_PARSER (AC3, ac3);
REGISTER_PARSER (ADX, adx);
REGISTER_PARSER (CAVSVIDEO, cavsvideo);
+ REGISTER_PARSER (COOK, cook);
REGISTER_PARSER (DCA, dca);
REGISTER_PARSER (DIRAC, dirac);
REGISTER_PARSER (DNXHD, dnxhd);
diff --git a/libavcodec/cook_parser.c b/libavcodec/cook_parser.c
new file mode 100644
index 0000000000..c16f7c8a5f
--- /dev/null
+++ b/libavcodec/cook_parser.c
@@ -0,0 +1,59 @@
+/*
+ * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
+ *
+ * This file is part of Libav.
+ *
+ * Libav 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.
+ *
+ * Libav 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 Libav; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * Cook audio parser
+ *
+ * Determines subpacket duration from extradata.
+ */
+
+#include <stdint.h>
+
+#include "libavutil/intreadwrite.h"
+#include "parser.h"
+
+typedef struct CookParseContext {
+ int duration;
+} CookParseContext;
+
+static int cook_parse(AVCodecParserContext *s1, AVCodecContext *avctx,
+ const uint8_t **poutbuf, int *poutbuf_size,
+ const uint8_t *buf, int buf_size)
+{
+ CookParseContext *s = s1->priv_data;
+
+ if (s->duration)
+ s1->duration = s->duration;
+ else if (avctx->extradata && avctx->extradata_size >= 8 && avctx->channels)
+ s->duration = AV_RB16(avctx->extradata + 4) / avctx->channels;
+
+ /* always return the full packet. this parser isn't doing any splitting or
+ combining, only setting packet duration */
+ *poutbuf = buf;
+ *poutbuf_size = buf_size;
+ return buf_size;
+}
+
+AVCodecParser ff_cook_parser = {
+ .codec_ids = { CODEC_ID_COOK },
+ .priv_data_size = sizeof(CookParseContext),
+ .parser_parse = cook_parse,
+};
diff --git a/libavcodec/version.h b/libavcodec/version.h
index 2429eb5e64..1a549ba63c 100644
--- a/libavcodec/version.h
+++ b/libavcodec/version.h
@@ -27,8 +27,8 @@
*/
#define LIBAVCODEC_VERSION_MAJOR 54
-#define LIBAVCODEC_VERSION_MINOR 14
-#define LIBAVCODEC_VERSION_MICRO 101
+#define LIBAVCODEC_VERSION_MINOR 15
+#define LIBAVCODEC_VERSION_MICRO 100
#define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
LIBAVCODEC_VERSION_MINOR, \
diff --git a/libavformat/daud.c b/libavformat/daud.c
index 2f74c3236d..916489fc0e 100644
--- a/libavformat/daud.c
+++ b/libavformat/daud.c
@@ -75,7 +75,7 @@ AVInputFormat ff_daud_demuxer = {
.long_name = NULL_IF_CONFIG_SMALL("D-Cinema audio format"),
.read_header = daud_header,
.read_packet = daud_packet,
- .extensions = "302",
+ .extensions = "302,daud",
};
#endif
diff --git a/libavformat/rmdec.c b/libavformat/rmdec.c
index bd83956c12..d919a9a41a 100644
--- a/libavformat/rmdec.c
+++ b/libavformat/rmdec.c
@@ -205,6 +205,7 @@ static int rm_read_audio_stream_info(AVFormatContext *s, AVIOContext *pb,
st->codec->block_align = coded_framesize;
break;
case CODEC_ID_COOK:
+ st->need_parsing = AVSTREAM_PARSE_HEADERS;
case CODEC_ID_ATRAC3:
case CODEC_ID_SIPR:
avio_rb16(pb); avio_r8(pb);
diff --git a/libavutil/Makefile b/libavutil/Makefile
index ed7de258ac..83e6c07605 100644
--- a/libavutil/Makefile
+++ b/libavutil/Makefile
@@ -5,6 +5,7 @@ NAME = avutil
HEADERS = adler32.h \
aes.h \
attributes.h \
+ audio_fifo.h \
audioconvert.h \
avassert.h \
avstring.h \
@@ -45,6 +46,7 @@ BUILT_HEADERS = avconfig.h
OBJS = adler32.o \
aes.o \
+ audio_fifo.o \
audioconvert.o \
avstring.o \
base64.o \
diff --git a/libavutil/audio_fifo.c b/libavutil/audio_fifo.c
new file mode 100644
index 0000000000..97c51a72c1
--- /dev/null
+++ b/libavutil/audio_fifo.c
@@ -0,0 +1,193 @@
+/*
+ * Audio FIFO
+ * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
+ *
+ * This file is part of Libav.
+ *
+ * Libav 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.
+ *
+ * Libav 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 Libav; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * Audio FIFO
+ */
+
+#include "avutil.h"
+#include "audio_fifo.h"
+#include "fifo.h"
+#include "mem.h"
+#include "samplefmt.h"
+
+struct AVAudioFifo {
+ AVFifoBuffer **buf; /**< single buffer for interleaved, per-channel buffers for planar */
+ int nb_buffers; /**< number of buffers */
+ int nb_samples; /**< number of samples currently in the FIFO */
+ int allocated_samples; /**< current allocated size, in samples */
+
+ int channels; /**< number of channels */
+ enum AVSampleFormat sample_fmt; /**< sample format */
+ int sample_size; /**< size, in bytes, of one sample in a buffer */
+};
+
+void av_audio_fifo_free(AVAudioFifo *af)
+{
+ if (af) {
+ if (af->buf) {
+ int i;
+ for (i = 0; i < af->nb_buffers; i++) {
+ if (af->buf[i])
+ av_fifo_free(af->buf[i]);
+ }
+ av_free(af->buf);
+ }
+ av_free(af);
+ }
+}
+
+AVAudioFifo *av_audio_fifo_alloc(enum AVSampleFormat sample_fmt, int channels,
+ int nb_samples)
+{
+ AVAudioFifo *af;
+ int buf_size, i;
+
+ /* get channel buffer size (also validates parameters) */
+ if (av_samples_get_buffer_size(&buf_size, channels, nb_samples, sample_fmt, 1) < 0)
+ return NULL;
+
+ af = av_mallocz(sizeof(*af));
+ if (!af)
+ return NULL;
+
+ af->channels = channels;
+ af->sample_fmt = sample_fmt;
+ af->sample_size = buf_size / nb_samples;
+ af->nb_buffers = av_sample_fmt_is_planar(sample_fmt) ? channels : 1;
+
+ af->buf = av_mallocz(af->nb_buffers * sizeof(*af->buf));
+ if (!af->buf)
+ goto error;
+
+ for (i = 0; i < af->nb_buffers; i++) {
+ af->buf[i] = av_fifo_alloc(buf_size);
+ if (!af->buf[i])
+ goto error;
+ }
+ af->allocated_samples = nb_samples;
+
+ return af;
+
+error:
+ av_audio_fifo_free(af);
+ return NULL;
+}
+
+int av_audio_fifo_realloc(AVAudioFifo *af, int nb_samples)
+{
+ int i, ret, buf_size;
+
+ if ((ret = av_samples_get_buffer_size(&buf_size, af->channels, nb_samples,
+ af->sample_fmt, 1)) < 0)
+ return ret;
+
+ for (i = 0; i < af->nb_buffers; i++) {
+ if ((ret = av_fifo_realloc2(af->buf[i], buf_size)) < 0)
+ return ret;
+ }
+ af->allocated_samples = nb_samples;
+ return 0;
+}
+
+int av_audio_fifo_write(AVAudioFifo *af, void **data, int nb_samples)
+{
+ int i, ret, size;
+
+ /* automatically reallocate buffers if needed */
+ if (av_audio_fifo_space(af) < nb_samples) {
+ int current_size = av_audio_fifo_size(af);
+ /* check for integer overflow in new size calculation */
+ if (INT_MAX / 2 - current_size < nb_samples)
+ return AVERROR(EINVAL);
+ /* reallocate buffers */
+ if ((ret = av_audio_fifo_realloc(af, 2 * (current_size + nb_samples))) < 0)
+ return ret;
+ }
+
+ size = nb_samples * af->sample_size;
+ for (i = 0; i < af->nb_buffers; i++) {
+ ret = av_fifo_generic_write(af->buf[i], data[i], size, NULL);
+ if (ret != size)
+ return AVERROR_BUG;
+ }
+ af->nb_samples += nb_samples;
+
+ return nb_samples;
+}
+
+int av_audio_fifo_read(AVAudioFifo *af, void **data, int nb_samples)
+{
+ int i, ret, size;
+
+ if (nb_samples < 0)
+ return AVERROR(EINVAL);
+ nb_samples = FFMIN(nb_samples, af->nb_samples);
+ if (!nb_samples)
+ return 0;
+
+ size = nb_samples * af->sample_size;
+ for (i = 0; i < af->nb_buffers; i++) {
+ if ((ret = av_fifo_generic_read(af->buf[i], data[i], size, NULL)) < 0)
+ return AVERROR_BUG;
+ }
+ af->nb_samples -= nb_samples;
+
+ return nb_samples;
+}
+
+int av_audio_fifo_drain(AVAudioFifo *af, int nb_samples)
+{
+ int i, size;
+
+ if (nb_samples < 0)
+ return AVERROR(EINVAL);
+ nb_samples = FFMIN(nb_samples, af->nb_samples);
+
+ if (nb_samples) {
+ size = nb_samples * af->sample_size;
+ for (i = 0; i < af->nb_buffers; i++)
+ av_fifo_drain(af->buf[i], size);
+ af->nb_samples -= nb_samples;
+ }
+ return 0;
+}
+
+void av_audio_fifo_reset(AVAudioFifo *af)
+{
+ int i;
+
+ for (i = 0; i < af->nb_buffers; i++)
+ av_fifo_reset(af->buf[i]);
+
+ af->nb_samples = 0;
+}
+
+int av_audio_fifo_size(AVAudioFifo *af)
+{
+ return af->nb_samples;
+}
+
+int av_audio_fifo_space(AVAudioFifo *af)
+{
+ return af->allocated_samples - af->nb_samples;
+}
diff --git a/libavutil/audio_fifo.h b/libavutil/audio_fifo.h
new file mode 100644
index 0000000000..8c76388255
--- /dev/null
+++ b/libavutil/audio_fifo.h
@@ -0,0 +1,146 @@
+/*
+ * Audio FIFO
+ * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
+ *
+ * This file is part of Libav.
+ *
+ * Libav 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.
+ *
+ * Libav 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 Libav; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * Audio FIFO Buffer
+ */
+
+#ifndef AVUTIL_AUDIO_FIFO_H
+#define AVUTIL_AUDIO_FIFO_H
+
+#include "avutil.h"
+#include "fifo.h"
+#include "samplefmt.h"
+
+/**
+ * @addtogroup lavu_audio
+ * @{
+ */
+
+/**
+ * Context for an Audio FIFO Buffer.
+ *
+ * - Operates at the sample level rather than the byte level.
+ * - Supports multiple channels with either planar or packed sample format.
+ * - Automatic reallocation when writing to a full buffer.
+ */
+typedef struct AVAudioFifo AVAudioFifo;
+
+/**
+ * Free an AVAudioFifo.
+ *
+ * @param af AVAudioFifo to free
+ */
+void av_audio_fifo_free(AVAudioFifo *af);
+
+/**
+ * Allocate an AVAudioFifo.
+ *
+ * @param sample_fmt sample format
+ * @param channels number of channels
+ * @param nb_samples initial allocation size, in samples
+ * @return newly allocated AVAudioFifo, or NULL on error
+ */
+AVAudioFifo *av_audio_fifo_alloc(enum AVSampleFormat sample_fmt, int channels,
+ int nb_samples);
+
+/**
+ * Reallocate an AVAudioFifo.
+ *
+ * @param af AVAudioFifo to reallocate
+ * @param nb_samples new allocation size, in samples
+ * @return 0 if OK, or negative AVERROR code on failure
+ */
+int av_audio_fifo_realloc(AVAudioFifo *af, int nb_samples);
+
+/**
+ * Write data to an AVAudioFifo.
+ *
+ * The AVAudioFifo will be reallocated automatically if the available space
+ * is less than nb_samples.
+ *
+ * @see enum AVSampleFormat
+ * The documentation for AVSampleFormat describes the data layout.
+ *
+ * @param af AVAudioFifo to write to
+ * @param data audio data plane pointers
+ * @param nb_samples number of samples to write
+ * @return number of samples actually written, or negative AVERROR
+ * code on failure.
+ */
+int av_audio_fifo_write(AVAudioFifo *af, void **data, int nb_samples);
+
+/**
+ * Read data from an AVAudioFifo.
+ *
+ * @see enum AVSampleFormat
+ * The documentation for AVSampleFormat describes the data layout.
+ *
+ * @param af AVAudioFifo to read from
+ * @param data audio data plane pointers
+ * @param nb_samples number of samples to read
+ * @return number of samples actually read, or negative AVERROR code
+ * on failure.
+ */
+int av_audio_fifo_read(AVAudioFifo *af, void **data, int nb_samples);
+
+/**
+ * Drain data from an AVAudioFifo.
+ *
+ * Removes the data without reading it.
+ *
+ * @param af AVAudioFifo to drain
+ * @param nb_samples number of samples to drain
+ * @return 0 if OK, or negative AVERROR code on failure
+ */
+int av_audio_fifo_drain(AVAudioFifo *af, int nb_samples);
+
+/**
+ * Reset the AVAudioFifo buffer.
+ *
+ * This empties all data in the buffer.
+ *
+ * @param af AVAudioFifo to reset
+ */
+void av_audio_fifo_reset(AVAudioFifo *af);
+
+/**
+ * Get the current number of samples in the AVAudioFifo available for reading.
+ *
+ * @param af the AVAudioFifo to query
+ * @return number of samples available for reading
+ */
+int av_audio_fifo_size(AVAudioFifo *af);
+
+/**
+ * Get the current number of samples in the AVAudioFifo available for writing.
+ *
+ * @param af the AVAudioFifo to query
+ * @return number of samples available for writing
+ */
+int av_audio_fifo_space(AVAudioFifo *af);
+
+/**
+ * @}
+ */
+
+#endif /* AVUTIL_AUDIO_FIFO_H */
diff --git a/libavutil/avutil.h b/libavutil/avutil.h
index 6464167473..0787bcbe09 100644
--- a/libavutil/avutil.h
+++ b/libavutil/avutil.h
@@ -153,7 +153,7 @@
*/
#define LIBAVUTIL_VERSION_MAJOR 51
-#define LIBAVUTIL_VERSION_MINOR 46
+#define LIBAVUTIL_VERSION_MINOR 47
#define LIBAVUTIL_VERSION_MICRO 100
#define LIBAVUTIL_VERSION_INT AV_VERSION_INT(LIBAVUTIL_VERSION_MAJOR, \
diff --git a/tests/Makefile b/tests/Makefile
index 1a94663c50..9b88c06f82 100644
--- a/tests/Makefile
+++ b/tests/Makefile
@@ -29,6 +29,9 @@ tests/data/asynth1.sw: tests/audiogen$(HOSTEXESUF) | tests/data
tests/data/asynth-16000-1.sw: tests/audiogen$(HOSTEXESUF) | tests/data
$(M)./$< $@ 16000 1
+tests/data/asynth-%.wav: tests/audiogen$(HOSTEXESUF) | tests/data
+ $(M)./$< $@ $(subst -, ,$*)
+
tests/data/mapchan-6ch.sw: tests/audiogen$(HOSTEXESUF)
@mkdir -p tests/data
$(M)./$< $@ 22050 6
@@ -37,7 +40,7 @@ tests/data/mapchan-mono.sw: tests/audiogen$(HOSTEXESUF)
@mkdir -p tests/data
$(M)./$< $@ 22050 1
-tests/data/%.sw tests/vsynth%/00.pgm: TAG = GEN
+tests/data/%.sw tests/data/asynth% tests/vsynth%/00.pgm: TAG = GEN
include $(SRC_PATH)/tests/fate/aac.mak
include $(SRC_PATH)/tests/fate/ac3.mak
diff --git a/tests/audiogen.c b/tests/audiogen.c
index 776fdf9316..5818797a65 100644
--- a/tests/audiogen.c
+++ b/tests/audiogen.c
@@ -22,7 +22,9 @@
*/
#include <stdlib.h>
+#include <stdint.h>
#include <stdio.h>
+#include <string.h>
#define MAX_CHANNELS 8
@@ -93,12 +95,45 @@ static int int_cos(int a)
FILE *outfile;
-static void put_sample(int v)
+static void put16(int16_t v)
{
- fputc(v & 0xff, outfile);
+ fputc( v & 0xff, outfile);
fputc((v >> 8) & 0xff, outfile);
}
+static void put32(uint32_t v)
+{
+ fputc( v & 0xff, outfile);
+ fputc((v >> 8) & 0xff, outfile);
+ fputc((v >> 16) & 0xff, outfile);
+ fputc((v >> 24) & 0xff, outfile);
+}
+
+#define HEADER_SIZE 46
+#define FMT_SIZE 18
+#define SAMPLE_SIZE 2
+#define WFORMAT_PCM 0x0001
+
+static void put_wav_header(int sample_rate, int channels, int nb_samples)
+{
+ int block_align = SAMPLE_SIZE * channels;
+ int data_size = block_align * nb_samples;
+
+ fputs("RIFF", outfile);
+ put32(HEADER_SIZE + data_size);
+ fputs("WAVEfmt ", outfile);
+ put32(FMT_SIZE);
+ put16(WFORMAT_PCM);
+ put16(channels);
+ put32(sample_rate);
+ put32(block_align * sample_rate);
+ put16(block_align);
+ put16(SAMPLE_SIZE * 8);
+ put16(0);
+ fputs("data", outfile);
+ put32(data_size);
+}
+
int main(int argc, char **argv)
{
int i, a, v, j, f, amp, ampa;
@@ -107,10 +142,12 @@ int main(int argc, char **argv)
int taba[MAX_CHANNELS];
int sample_rate = 44100;
int nb_channels = 2;
+ char *ext;
if (argc < 2 || argc > 4) {
printf("usage: %s file [<sample rate> [<channels>]]\n"
"generate a test raw 16 bit audio stream\n"
+ "If the file extension is .wav a WAVE header will be added.\n"
"default: 44100 Hz stereo\n", argv[0]);
exit(1);
}
@@ -137,12 +174,15 @@ int main(int argc, char **argv)
return 1;
}
+ if ((ext = strrchr(argv[1], '.')) != NULL && !strcmp(ext, ".wav"))
+ put_wav_header(sample_rate, nb_channels, 6 * sample_rate);
+
/* 1 second of single freq sinus at 1000 Hz */
a = 0;
for (i = 0; i < 1 * sample_rate; i++) {
v = (int_cos(a) * 10000) >> FRAC_BITS;
for (j = 0; j < nb_channels; j++)
- put_sample(v);
+ put16(v);
a += (1000 * FRAC_ONE) / sample_rate;
}
@@ -151,7 +191,7 @@ int main(int argc, char **argv)
for (i = 0; i < 1 * sample_rate; i++) {
v = (int_cos(a) * 10000) >> FRAC_BITS;
for (j = 0; j < nb_channels; j++)
- put_sample(v);
+ put16(v);
f = 100 + (((10000 - 100) * i) / sample_rate);
a += (f * FRAC_ONE) / sample_rate;
}
@@ -160,14 +200,14 @@ int main(int argc, char **argv)
for (i = 0; i < sample_rate / 2; i++) {
v = myrnd(&seed, 20000) - 10000;
for (j = 0; j < nb_channels; j++)
- put_sample(v);
+ put16(v);
}
/* 0.5 second of high amplitude white noise */
for (i = 0; i < sample_rate / 2; i++) {
v = myrnd(&seed, 65535) - 32768;
for (j = 0; j < nb_channels; j++)
- put_sample(v);
+ put16(v);
}
/* 1 second of unrelated ramps for each channel */
@@ -179,7 +219,7 @@ int main(int argc, char **argv)
for (i = 0; i < 1 * sample_rate; i++) {
for (j = 0; j < nb_channels; j++) {
v = (int_cos(taba[j]) * 10000) >> FRAC_BITS;
- put_sample(v);
+ put16(v);
f = tabf1[j] + (((tabf2[j] - tabf1[j]) * i) / sample_rate);
taba[j] += (f * FRAC_ONE) / sample_rate;
}
@@ -194,7 +234,7 @@ int main(int argc, char **argv)
if (j & 1)
amp = 10000 - amp;
v = (int_cos(a) * amp) >> FRAC_BITS;
- put_sample(v);
+ put16(v);
a += (500 * FRAC_ONE) / sample_rate;
ampa += (2 * FRAC_ONE) / sample_rate;
}
diff --git a/tests/codec-regression.sh b/tests/codec-regression.sh
index 1e5f8166bd..51d1b4c67b 100755
--- a/tests/codec-regression.sh
+++ b/tests/codec-regression.sh
@@ -395,16 +395,6 @@ do_audio_encoding g723_1.tco "-b:a 6.3k -ac 1 -ar 8000 -acodec g723_1"
do_audio_decoding
fi
-if [ -n "$do_g722" ] ; then
-do_audio_encoding g722.wav "-b 64k -ac 1 -ar 16000 -acodec g722"
-do_audio_decoding
-fi
-
-if [ -n "$do_g726" ] ; then
-do_audio_encoding g726.wav "-b:a 32k -ac 1 -ar 8000 -acodec g726"
-do_audio_decoding
-fi
-
if [ -n "$do_adpcm_adx" ] ; then
do_audio_encoding adpcm_adx.adx "-acodec adpcm_adx"
do_audio_decoding
@@ -534,6 +524,3 @@ fi
if [ -n "$do_pcm_f64le" ] ; then
do_audio_enc_dec wav dbl pcm_f64le
fi
-if [ -n "$do_pcm_s24daud" ] ; then
-do_audio_enc_dec 302 s16 pcm_s24daud "-ac 6 -ar 96000"
-fi
diff --git a/tests/fate/pcm.mak b/tests/fate/pcm.mak
index 3ce04f9cea..4b271346f6 100644
--- a/tests/fate/pcm.mak
+++ b/tests/fate/pcm.mak
@@ -25,5 +25,10 @@ fate-pcm_u8-stereo: CMD = md5 -i $(SAMPLES)/qt-surge-suite/surge-2-8-raw.mov -f
FATE_PCM += fate-w64
fate-w64: CMD = crc -i $(SAMPLES)/w64/w64-pcm16.w64
+FATE_PCM += fate-dcinema-encode
+fate-dcinema-encode: tests/data/asynth-96000-6.wav
+fate-dcinema-encode: SRC = tests/data/asynth-96000-6.wav
+fate-dcinema-encode: CMD = enc_dec_pcm daud md5 s16le $(SRC) -c:a pcm_s24daud
+
FATE_TESTS += $(FATE_PCM)
fate-pcm: $(FATE_PCM)
diff --git a/tests/fate/voice.mak b/tests/fate/voice.mak
index b9bf8e1e51..f8fa24b7b6 100644
--- a/tests/fate/voice.mak
+++ b/tests/fate/voice.mak
@@ -1,9 +1,36 @@
-FATE_VOICE += fate-g722dec-1
+FATE_G722 += fate-g722dec-1
fate-g722dec-1: CMD = framecrc -i $(SAMPLES)/g722/conf-adminmenu-162.g722
-FATE_VOICE += fate-g722enc
-fate-g722enc: tests/data/asynth-16000-1.sw
-fate-g722enc: CMD = md5 -ar 16000 -ac 1 -f s16le -i $(TARGET_PATH)/tests/data/asynth-16000-1.sw -acodec g722 -ac 1 -f g722
+FATE_G722 += fate-g722-encode
+fate-g722-encode: tests/data/asynth-16000-1.wav
+fate-g722-encode: SRC = tests/data/asynth-16000-1.wav
+fate-g722-encode: CMD = enc_dec_pcm wav md5 s16le $(SRC) -c:a g722
+
+FATE_VOICE += $(FATE_G722)
+fate-g722: $(FATE_G722)
+
+FATE_G726 += fate-g726-encode-2bit
+fate-g726-encode-2bit: tests/data/asynth-8000-1.wav
+fate-g726-encode-2bit: SRC = tests/data/asynth-8000-1.wav
+fate-g726-encode-2bit: CMD = enc_dec_pcm wav md5 s16le $(SRC) -c:a g726 -b:a 16k
+
+FATE_G726 += fate-g726-encode-3bit
+fate-g726-encode-3bit: tests/data/asynth-8000-1.wav
+fate-g726-encode-3bit: SRC = tests/data/asynth-8000-1.wav
+fate-g726-encode-3bit: CMD = enc_dec_pcm wav md5 s16le $(SRC) -c:a g726 -b:a 24k
+
+FATE_G726 += fate-g726-encode-4bit
+fate-g726-encode-4bit: tests/data/asynth-8000-1.wav
+fate-g726-encode-4bit: SRC = tests/data/asynth-8000-1.wav
+fate-g726-encode-4bit: CMD = enc_dec_pcm wav md5 s16le $(SRC) -c:a g726 -b:a 32k
+
+FATE_G726 += fate-g726-encode-5bit
+fate-g726-encode-5bit: tests/data/asynth-8000-1.wav
+fate-g726-encode-5bit: SRC = tests/data/asynth-8000-1.wav
+fate-g726-encode-5bit: CMD = enc_dec_pcm wav md5 s16le $(SRC) -c:a g726 -b:a 40k
+
+FATE_VOICE += $(FATE_G726)
+fate-g726: $(FATE_G726)
FATE_GSM += fate-gsm-ms
fate-gsm-ms: CMD = framecrc -i $(SAMPLES)/gsm/ciao.wav
diff --git a/tests/lavf-regression.sh b/tests/lavf-regression.sh
index 24a0c64ce8..2d27519fa1 100755
--- a/tests/lavf-regression.sh
+++ b/tests/lavf-regression.sh
@@ -24,18 +24,18 @@ do_lavf_fate()
do_lavf()
{
file=${outfile}lavf.$1
- do_avconv $file $DEC_OPTS -f image2 -vcodec pgmyuv -i $raw_src $DEC_OPTS -ar 44100 -f s16le -i $pcm_src $ENC_OPTS -b:a 64k -t 1 -qscale:v 10 $2
- do_avconv_crc $file $DEC_OPTS -i $target_path/$file $3
+ do_avconv $file $DEC_OPTS -f image2 -vcodec pgmyuv -i $raw_src $DEC_OPTS -ar 44100 -f s16le $2 -i $pcm_src $ENC_OPTS -b:a 64k -t 1 -qscale:v 10 $3
+ do_avconv_crc $file $DEC_OPTS -i $target_path/$file $4
}
-do_lavf_timecode_nodrop() { do_lavf $1 "$2 -timecode 02:56:14:13"; }
-do_lavf_timecode_drop() { do_lavf $1 "$2 -timecode 02:56:14.13 -r 30000/1001"; }
+do_lavf_timecode_nodrop() { do_lavf $1 "" "$2 -timecode 02:56:14:13"; }
+do_lavf_timecode_drop() { do_lavf $1 "" "$2 -timecode 02:56:14.13 -r 30000/1001"; }
do_lavf_timecode()
{
do_lavf_timecode_nodrop "$@"
do_lavf_timecode_drop "$@"
- do_lavf "$@"
+ do_lavf $1 "" "$2"
}
do_streamed_images()
@@ -64,11 +64,11 @@ do_audio_only()
}
if [ -n "$do_avi" ] ; then
-do_lavf avi "-acodec mp2 -ab 64k"
+do_lavf avi "" "-acodec mp2 -ab 64k"
fi
if [ -n "$do_asf" ] ; then
-do_lavf asf "-acodec mp2 -ab 64k" "-r 25"
+do_lavf asf "" "-acodec mp2 -ab 64k" "-r 25"
fi
if [ -n "$do_rm" ] ; then
@@ -87,15 +87,15 @@ do_lavf_timecode mxf "-ar 48000 -bf 2"
fi
if [ -n "$do_mxf_d10" ]; then
-do_lavf mxf_d10 "-ar 48000 -ac 2 -r 25 -s 720x576 -vf pad=720:608:0:32 -vcodec mpeg2video -g 0 -flags +ildct+low_delay -dc 10 -non_linear_quant 1 -intra_vlc 1 -qscale 1 -ps 1 -qmin 1 -rc_max_vbv_use 1 -rc_min_vbv_use 1 -pix_fmt yuv422p -minrate 30000k -maxrate 30000k -b 30000k -bufsize 1200000 -top 1 -rc_init_occupancy 1200000 -qmax 12 -f mxf_d10"
+do_lavf mxf_d10 "-ar 48000 -ac 2" "-r 25 -s 720x576 -vf pad=720:608:0:32 -vcodec mpeg2video -g 0 -flags +ildct+low_delay -dc 10 -non_linear_quant 1 -intra_vlc 1 -qscale 1 -ps 1 -qmin 1 -rc_max_vbv_use 1 -rc_min_vbv_use 1 -pix_fmt yuv422p -minrate 30000k -maxrate 30000k -b 30000k -bufsize 1200000 -top 1 -rc_init_occupancy 1200000 -qmax 12 -f mxf_d10"
fi
if [ -n "$do_ts" ] ; then
-do_lavf ts "-ab 64k -mpegts_transport_stream_id 42"
+do_lavf ts "" "-ab 64k -mpegts_transport_stream_id 42"
fi
if [ -n "$do_swf" ] ; then
-do_lavf swf -an
+do_lavf swf "" "-an"
fi
if [ -n "$do_ffm" ] ; then
@@ -103,11 +103,11 @@ do_lavf ffm "-ab 64k"
fi
if [ -n "$do_flv_fmt" ] ; then
-do_lavf flv -an
+do_lavf flv "" "-an"
fi
if [ -n "$do_mov" ] ; then
-do_lavf mov "-movflags +rtphint -acodec pcm_alaw -vcodec mpeg4"
+do_lavf mov "" "-movflags +rtphint -acodec pcm_alaw -vcodec mpeg4"
do_lavf_timecode mov "-acodec pcm_alaw -vcodec mpeg4"
fi
@@ -118,21 +118,21 @@ fi
if [ -n "$do_dv_fmt" ] ; then
do_lavf_timecode_nodrop dv "-ar 48000 -r 25 -s pal -ac 2"
do_lavf_timecode_drop dv "-ar 48000 -pix_fmt yuv411p -s ntsc -ac 2"
-do_lavf dv "-ar 48000 -r 25 -s pal -ac 2"
+do_lavf dv "-ar 48000" "-r 25 -s pal -ac 2"
fi
if [ -n "$do_gxf" ] ; then
do_lavf_timecode_nodrop gxf "-ar 48000 -r 25 -s pal -ac 1"
do_lavf_timecode_drop gxf "-ar 48000 -s ntsc -ac 1"
-do_lavf gxf "-ar 48000 -r 25 -s pal -ac 1"
+do_lavf gxf "-ar 48000" "-r 25 -s pal -ac 1"
fi
if [ -n "$do_nut" ] ; then
-do_lavf nut "-acodec mp2 -ab 64k"
+do_lavf nut "" "-acodec mp2 -ab 64k"
fi
if [ -n "$do_mkv" ] ; then
-do_lavf mkv "-acodec mp2 -ab 64k -vcodec mpeg4"
+do_lavf mkv "" "-acodec mp2 -ab 64k -vcodec mpeg4"
fi
if [ -n "$do_mp3" ] ; then
@@ -150,7 +150,7 @@ do_lavf_fate ogg "vp3/coeff_level64.mkv"
fi
if [ -n "$do_wtv" ] ; then
-do_lavf wtv "-acodec mp2"
+do_lavf wtv "" "-acodec mp2"
fi
diff --git a/tests/ref/acodec/g722 b/tests/ref/acodec/g722
deleted file mode 100644
index de0e5a0caa..0000000000
--- a/tests/ref/acodec/g722
+++ /dev/null
@@ -1,4 +0,0 @@
-e4d5ae038f29659c03fcf68818f7be6c *./tests/data/acodec/g722.wav
-48053 ./tests/data/acodec/g722.wav
-8dafe5b74ccd5f08fed2fb2a69c5475f *./tests/data/g722.acodec.out.wav
-stddev: 8939.47 PSNR: 17.30 MAXDIFF:40370 bytes: 191980/ 1058400
diff --git a/tests/ref/acodec/g726 b/tests/ref/acodec/g726
deleted file mode 100644
index c519ac297e..0000000000
--- a/tests/ref/acodec/g726
+++ /dev/null
@@ -1,4 +0,0 @@
-331fcf91f4483b508059d0933af97987 *./tests/data/acodec/g726.wav
-24054 ./tests/data/acodec/g726.wav
-fac563ba7947d8fc42b4af048707c145 *./tests/data/g726.acodec.out.wav
-stddev: 8553.69 PSNR: 17.69 MAXDIFF:29353 bytes: 95984/ 1058400
diff --git a/tests/ref/acodec/pcm_s24daud b/tests/ref/acodec/pcm_s24daud
deleted file mode 100644
index e1f22964fc..0000000000
--- a/tests/ref/acodec/pcm_s24daud
+++ /dev/null
@@ -1,4 +0,0 @@
-1b75d5198ae789ab3c48f7024e08f4a9 *./tests/data/acodec/pcm_s24daud.302
-10368730 ./tests/data/acodec/pcm_s24daud.302
-70ec0ba6bc151ddc7509c09804d95d3b *./tests/data/pcm_s24daud.acodec.out.wav
-stddev: 8967.92 PSNR: 17.28 MAXDIFF:42548 bytes: 6911796/ 1058400
diff --git a/tests/ref/fate/dcinema-encode b/tests/ref/fate/dcinema-encode
new file mode 100644
index 0000000000..8aeb21526c
--- /dev/null
+++ b/tests/ref/fate/dcinema-encode
@@ -0,0 +1 @@
+MD5=2d7c6897c315493647db159f4bfd6edc
diff --git a/tests/ref/fate/g722-encode b/tests/ref/fate/g722-encode
new file mode 100644
index 0000000000..c7198cf83c
--- /dev/null
+++ b/tests/ref/fate/g722-encode
@@ -0,0 +1 @@
+MD5=7106189574186051c0497b287e2e5f19
diff --git a/tests/ref/fate/g722enc b/tests/ref/fate/g722enc
deleted file mode 100644
index 9b8e469a8b..0000000000
--- a/tests/ref/fate/g722enc
+++ /dev/null
@@ -1 +0,0 @@
-94e2f200d6e05b47cec4aa3e94571cf3
diff --git a/tests/ref/fate/g726-encode-2bit b/tests/ref/fate/g726-encode-2bit
new file mode 100644
index 0000000000..26a12190fc
--- /dev/null
+++ b/tests/ref/fate/g726-encode-2bit
@@ -0,0 +1 @@
+MD5=215eaef5778a16e2bf4f3725a557f355
diff --git a/tests/ref/fate/g726-encode-3bit b/tests/ref/fate/g726-encode-3bit
new file mode 100644
index 0000000000..f9c6940217
--- /dev/null
+++ b/tests/ref/fate/g726-encode-3bit
@@ -0,0 +1 @@
+MD5=0bebd949dfd5ac0ae3f2c3ceb2e3fac1
diff --git a/tests/ref/fate/g726-encode-4bit b/tests/ref/fate/g726-encode-4bit
new file mode 100644
index 0000000000..6d03517164
--- /dev/null
+++ b/tests/ref/fate/g726-encode-4bit
@@ -0,0 +1 @@
+MD5=a21cfea116ab2179eabe5d84b6bfc09a
diff --git a/tests/ref/fate/g726-encode-5bit b/tests/ref/fate/g726-encode-5bit
new file mode 100644
index 0000000000..459ebb39f2
--- /dev/null
+++ b/tests/ref/fate/g726-encode-5bit
@@ -0,0 +1 @@
+MD5=9cad98cf5205bf76d6e9d1241e56141a
diff --git a/tests/ref/lavf/dv_fmt b/tests/ref/lavf/dv_fmt
index 884c079243..ea667b9627 100644
--- a/tests/ref/lavf/dv_fmt
+++ b/tests/ref/lavf/dv_fmt
@@ -4,6 +4,6 @@
cc33ae4f9e6828914dea0f09d1241b7e *./tests/data/lavf/lavf.dv
3480000 ./tests/data/lavf/lavf.dv
./tests/data/lavf/lavf.dv CRC=0x8d5e9e8f
-3a6a9163a67b729b4a6b5d972ccceb97 *./tests/data/lavf/lavf.dv
+b36c83cd0ba0ebe719f09f885c4bbcd3 *./tests/data/lavf/lavf.dv
3600000 ./tests/data/lavf/lavf.dv
-./tests/data/lavf/lavf.dv CRC=0x5ce4e5e4
+./tests/data/lavf/lavf.dv CRC=0x2bc2ae3a
diff --git a/tests/ref/lavf/gxf b/tests/ref/lavf/gxf
index 5796895a1d..63101b89ed 100644
--- a/tests/ref/lavf/gxf
+++ b/tests/ref/lavf/gxf
@@ -4,6 +4,6 @@ befc1a39c37a4ecd9264942a3e34b3f6 *./tests/data/lavf/lavf.gxf
267d2b2b6e357209d76c366302cf35c3 *./tests/data/lavf/lavf.gxf
794572 ./tests/data/lavf/lavf.gxf
./tests/data/lavf/lavf.gxf CRC=0xab47d02d
-1c1693cf2358025f1e37ac76e1da925a *./tests/data/lavf/lavf.gxf
+0a1a37fa79b62435545271b4e8e882f5 *./tests/data/lavf/lavf.gxf
796392 ./tests/data/lavf/lavf.gxf
-./tests/data/lavf/lavf.gxf CRC=0x102918fd
+./tests/data/lavf/lavf.gxf CRC=0x3b1a8e91
diff --git a/tests/ref/lavf/mxf_d10 b/tests/ref/lavf/mxf_d10
index efeac5a2c0..a10e8456ed 100644
--- a/tests/ref/lavf/mxf_d10
+++ b/tests/ref/lavf/mxf_d10
@@ -1,3 +1,3 @@
-23177c8a72f34e243e9ffc4f6c70d3c7 *./tests/data/lavf/lavf.mxf_d10
+0d72247067569901a2e351586ddc0b82 *./tests/data/lavf/lavf.mxf_d10
5330989 ./tests/data/lavf/lavf.mxf_d10
-./tests/data/lavf/lavf.mxf_d10 CRC=0x81602ff1
+./tests/data/lavf/lavf.mxf_d10 CRC=0x4474d480
diff --git a/tests/ref/seek/g726_wav b/tests/ref/seek/g726_wav
deleted file mode 100644
index 0e145a2722..0000000000
--- a/tests/ref/seek/g726_wav
+++ /dev/null
@@ -1,53 +0,0 @@
-ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 58 size: 4096
-ret: 0 st:-1 flags:0 ts:-1.000000
-ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 58 size: 4096
-ret: 0 st:-1 flags:1 ts: 1.894167
-ret: 0 st: 0 flags:1 dts: 1.894000 pts: 1.894000 pos: 7634 size: 4096
-ret: 0 st: 0 flags:0 ts: 0.788375
-ret: 0 st: 0 flags:1 dts: 0.788500 pts: 0.788500 pos: 3212 size: 4096
-ret: 0 st: 0 flags:1 ts:-0.317500
-ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 58 size: 4096
-ret: 0 st:-1 flags:0 ts: 2.576668
-ret: 0 st: 0 flags:1 dts: 2.576750 pts: 2.576750 pos: 10365 size: 4096
-ret: 0 st:-1 flags:1 ts: 1.470835
-ret: 0 st: 0 flags:1 dts: 1.470750 pts: 1.470750 pos: 5941 size: 4096
-ret: 0 st: 0 flags:0 ts: 0.365000
-ret: 0 st: 0 flags:1 dts: 0.365000 pts: 0.365000 pos: 1518 size: 4096
-ret: 0 st: 0 flags:1 ts:-0.740875
-ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 58 size: 4096
-ret: 0 st:-1 flags:0 ts: 2.153336
-ret: 0 st: 0 flags:1 dts: 2.153500 pts: 2.153500 pos: 8672 size: 4096
-ret: 0 st:-1 flags:1 ts: 1.047503
-ret: 0 st: 0 flags:1 dts: 1.047500 pts: 1.047500 pos: 4248 size: 4096
-ret: 0 st: 0 flags:0 ts:-0.058375
-ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 58 size: 4096
-ret: 0 st: 0 flags:1 ts: 2.835875
-ret: 0 st: 0 flags:1 dts: 2.835750 pts: 2.835750 pos: 11401 size: 4096
-ret: 0 st:-1 flags:0 ts: 1.730004
-ret: 0 st: 0 flags:1 dts: 1.730000 pts: 1.730000 pos: 6978 size: 4096
-ret: 0 st:-1 flags:1 ts: 0.624171
-ret: 0 st: 0 flags:1 dts: 0.624000 pts: 0.624000 pos: 2554 size: 4096
-ret: 0 st: 0 flags:0 ts:-0.481625
-ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 58 size: 4096
-ret: 0 st: 0 flags:1 ts: 2.412500
-ret: 0 st: 0 flags:1 dts: 2.412500 pts: 2.412500 pos: 9708 size: 4096
-ret: 0 st:-1 flags:0 ts: 1.306672
-ret: 0 st: 0 flags:1 dts: 1.306750 pts: 1.306750 pos: 5285 size: 4096
-ret: 0 st:-1 flags:1 ts: 0.200839
-ret: 0 st: 0 flags:1 dts: 0.200750 pts: 0.200750 pos: 861 size: 4096
-ret: 0 st: 0 flags:0 ts:-0.905000
-ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 58 size: 4096
-ret: 0 st: 0 flags:1 ts: 1.989125
-ret: 0 st: 0 flags:1 dts: 1.989000 pts: 1.989000 pos: 8014 size: 4096
-ret: 0 st:-1 flags:0 ts: 0.883340
-ret: 0 st: 0 flags:1 dts: 0.883500 pts: 0.883500 pos: 3592 size: 4096
-ret: 0 st:-1 flags:1 ts:-0.222493
-ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 58 size: 4096
-ret: 0 st: 0 flags:0 ts: 2.671625
-ret: 0 st: 0 flags:1 dts: 2.671750 pts: 2.671750 pos: 10745 size: 4096
-ret: 0 st: 0 flags:1 ts: 1.565875
-ret: 0 st: 0 flags:1 dts: 1.565750 pts: 1.565750 pos: 6321 size: 4096
-ret: 0 st:-1 flags:0 ts: 0.460008
-ret: 0 st: 0 flags:1 dts: 0.460000 pts: 0.460000 pos: 1898 size: 4096
-ret: 0 st:-1 flags:1 ts:-0.645825
-ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 58 size: 4096
diff --git a/tests/ref/seek/pcm_s24daud_302 b/tests/ref/seek/pcm_s24daud_302
deleted file mode 100644
index 5c9b6d976b..0000000000
--- a/tests/ref/seek/pcm_s24daud_302
+++ /dev/null
@@ -1,27 +0,0 @@
-ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 4 size: 39816
-ret:-1 st:-1 flags:0 ts:-1.000000
-ret:-1 st:-1 flags:1 ts: 1.894167
-ret:-1 st: 0 flags:0 ts: 0.788333
-ret:-1 st: 0 flags:1 ts:-0.317500
-ret:-1 st:-1 flags:0 ts: 2.576668
-ret:-1 st:-1 flags:1 ts: 1.470835
-ret:-1 st: 0 flags:0 ts: 0.365000
-ret:-1 st: 0 flags:1 ts:-0.740833
-ret:-1 st:-1 flags:0 ts: 2.153336
-ret:-1 st:-1 flags:1 ts: 1.047503
-ret:-1 st: 0 flags:0 ts:-0.058333
-ret:-1 st: 0 flags:1 ts: 2.835833
-ret:-1 st:-1 flags:0 ts: 1.730004
-ret:-1 st:-1 flags:1 ts: 0.624171
-ret:-1 st: 0 flags:0 ts:-0.481667
-ret:-1 st: 0 flags:1 ts: 2.412500
-ret:-1 st:-1 flags:0 ts: 1.306672
-ret:-1 st:-1 flags:1 ts: 0.200839
-ret:-1 st: 0 flags:0 ts:-0.904989
-ret:-1 st: 0 flags:1 ts: 1.989178
-ret:-1 st:-1 flags:0 ts: 0.883340
-ret:-1 st:-1 flags:1 ts:-0.222493
-ret:-1 st: 0 flags:0 ts: 2.671678
-ret:-1 st: 0 flags:1 ts: 1.565844
-ret:-1 st:-1 flags:0 ts: 0.460008
-ret:-1 st:-1 flags:1 ts:-0.645825