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>

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()
 

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

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);

libavcodec/cook_parser.c

+/*
+ * 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,
+};

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, \

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
 

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);

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                                                         \

libavutil/audio_fifo.c

+/*
+ * 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;
+}

libavutil/audio_fifo.h

+/*
+ * 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 */

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, \

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

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;
         }

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

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)

tests/fate/voice.mak

-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

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
 
 

tests/ref/acodec/g722

-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

tests/ref/acodec/g726

-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

tests/ref/acodec/pcm_s24daud

-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

tests/ref/fate/dcinema-encode

+MD5=2d7c6897c315493647db159f4bfd6edc

tests/ref/fate/g722-encode

+MD5=7106189574186051c0497b287e2e5f19

tests/ref/fate/g722enc

-94e2f200d6e05b47cec4aa3e94571cf3

tests/ref/fate/g726-encode-2bit

+MD5=215eaef5778a16e2bf4f3725a557f355

tests/ref/fate/g726-encode-3bit

+MD5=0bebd949dfd5ac0ae3f2c3ceb2e3fac1

tests/ref/fate/g726-encode-4bit

+MD5=a21cfea116ab2179eabe5d84b6bfc09a

tests/ref/fate/g726-encode-5bit

+MD5=9cad98cf5205bf76d6e9d1241e56141a

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

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

tests/ref/lavf/mxf_d10

-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

tests/ref/seek/g726_wav

-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

tests/ref/seek/pcm_s24daud_302

-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