@@ -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()

@@ -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

@@ -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);

new file mode 100644

@@ -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,

+};

@@ -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, \

@@ -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

@@ -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);

@@ -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                                                         \

new file mode 100644

@@ -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;

+}

new file mode 100644

@@ -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 */

@@ -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, \

@@ -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

@@ -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;

         }

@@ -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

@@ -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)

@@ -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

@@ -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

deleted file mode 100644

@@ -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

deleted file mode 100644

@@ -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

deleted file mode 100644

@@ -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

new file mode 100644

@@ -0,0 +1 @@

+MD5=2d7c6897c315493647db159f4bfd6edc

new file mode 100644

@@ -0,0 +1 @@

+MD5=7106189574186051c0497b287e2e5f19