rename libeplayer3-arm -> libplayer3

Origin commit data
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Branch: master
Commit: e1b78787fd
Author: max_10 <max_10@gmx.de>
Date: 2019-06-15 (Sat, 15 Jun 2019)

Origin message was:
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- rename libeplayer3-arm -> libplayer3

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No further description and justification available within origin commit message!

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This commit was generated by Migit

libeplayer3/external/ffmpeg/src/bitstream.c

@@ -0,0 +1,78 @@
+/*
+ * Common bit i/o utils
+ * Copyright (c) 2000, 2001 Fabrice Bellard
+ * Copyright (c) 2002-2004 Michael Niedermayer <michaelni@gmx.at>
+ * Copyright (c) 2010 Loren Merritt
+ *
+ * alternative bitstream reader & writer by Michael Niedermayer <michaelni@gmx.at>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg 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.
+ *
+ * FFmpeg 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 FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * bitstream api.
+ */
+
+#include <libavutil/avassert.h>
+#include <ffmpeg/put_bits.h>
+
+
+void avpriv_align_put_bits(PutBitContext *s)
+{
+	put_bits(s, s->bit_left & 7, 0);
+}
+
+void avpriv_put_string(PutBitContext *pb, const char *string, int terminate_string)
+{
+	while (*string)
+	{
+		put_bits(pb, 8, *string);
+		string++;
+	}
+	if (terminate_string)
+		put_bits(pb, 8, 0);
+}
+
+void avpriv_copy_bits(PutBitContext *pb, const uint8_t *src, int length)
+{
+	int words = length >> 4;
+	int bits  = length & 15;
+	int i;
+
+	if (length == 0)
+		return;
+
+	av_assert0(length <= put_bits_left(pb));
+
+	if (words < 16 || put_bits_count(pb) & 7)
+	{
+		for (i = 0; i < words; i++)
+			put_bits(pb, 16, AV_RB16(src + 2 * i));
+	}
+	else
+	{
+		for (i = 0; put_bits_count(pb) & 31; i++)
+			put_bits(pb, 8, src[i]);
+		flush_put_bits(pb);
+		memcpy(put_bits_ptr(pb), src + i, 2 * words - i);
+		skip_put_bytes(pb, 2 * words - i);
+	}
+
+	put_bits(pb, bits, AV_RB16(src + 2 * words) >> (16 - bits));
+}
+

libeplayer3/external/ffmpeg/src/latmenc.c

@@ -0,0 +1,171 @@
+/*
+ * LATM/LOAS muxer
+ * Copyright (c) 2011 Kieran Kunhya <kieran@kunhya.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg 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.
+ *
+ * FFmpeg 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 FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/* ***************************** */
+/* Includes                      */
+/* ***************************** */
+#include <ffmpeg/get_bits.h>
+#include <ffmpeg/put_bits.h>
+#include <ffmpeg/mpeg4audio.h>
+#include <ffmpeg/latmenc.h>
+#include "debug.h"
+
+/* ***************************** */
+/* Makros/Constants              */
+/* ***************************** */
+
+int latmenc_decode_extradata(LATMContext *ctx, uint8_t *buf, int size)
+{
+	MPEG4AudioConfig m4ac;
+
+	if (size > MAX_EXTRADATA_SIZE)
+	{
+		latmenc_err("Extradata is larger than currently supported.\n");
+		return AVERROR_INVALIDDATA;
+	}
+	ctx->off = avpriv_mpeg4audio_get_config(&m4ac, buf, size * 8, 1);
+	if (ctx->off < 0)
+		return ctx->off;
+
+	if (ctx->object_type == AOT_ALS && (ctx->off & 7))
+	{
+		// as long as avpriv_mpeg4audio_get_config works correctly this is impossible
+		latmenc_err("BUG: ALS offset is not byte-aligned\n");
+		return AVERROR_INVALIDDATA;
+	}
+	/* FIXME: are any formats not allowed in LATM? */
+
+	if (m4ac.object_type > AOT_SBR && m4ac.object_type != AOT_ALS)
+	{
+		latmenc_err("Muxing MPEG-4 AOT %d in LATM is not supported\n", m4ac.object_type);
+		return AVERROR_INVALIDDATA;
+	}
+	ctx->channel_conf = m4ac.chan_config;
+	ctx->object_type  = m4ac.object_type;
+
+	return 0;
+}
+
+static void latmenc_write_frame_header(LATMContext *ctx, uint8_t *extradata, int extradata_size, PutBitContext *bs)
+{
+	int header_size;
+
+	/* AudioMuxElement */
+	put_bits(bs, 1, !!ctx->counter);
+
+	if (!ctx->counter)
+	{
+		/* StreamMuxConfig */
+		put_bits(bs, 1, 0); /* audioMuxVersion */
+		put_bits(bs, 1, 1); /* allStreamsSameTimeFraming */
+		put_bits(bs, 6, 0); /* numSubFrames */
+		put_bits(bs, 4, 0); /* numProgram */
+		put_bits(bs, 3, 0); /* numLayer */
+
+		/* AudioSpecificConfig */
+		if (ctx->object_type == AOT_ALS)
+		{
+			header_size = extradata_size - (ctx->off >> 3);
+			avpriv_copy_bits(bs, &extradata[ctx->off >> 3], header_size);
+		}
+		else
+		{
+			// + 3 assumes not scalable and dependsOnCoreCoder == 0,
+			// see decode_ga_specific_config in libavcodec/aacdec.c
+			avpriv_copy_bits(bs, extradata, ctx->off + 3);
+
+			if (!ctx->channel_conf)
+			{
+				GetBitContext gb;
+				int ret = init_get_bits8(&gb, extradata, extradata_size);
+				av_assert0(ret >= 0); // extradata size has been checked already, so this should not fail
+				skip_bits_long(&gb, ctx->off + 3);
+				avpriv_copy_pce_data(bs, &gb);
+			}
+		}
+
+		put_bits(bs, 3, 0); /* frameLengthType */
+		put_bits(bs, 8, 0xff); /* latmBufferFullness */
+
+		put_bits(bs, 1, 0); /* otherDataPresent */
+		put_bits(bs, 1, 0); /* crcCheckPresent */
+	}
+
+	ctx->counter++;
+	ctx->counter %= ctx->mod;
+}
+
+int latmenc_write_packet(LATMContext *ctx, uint8_t *data, int size, uint8_t *extradata, int extradata_size)
+{
+	PutBitContext bs;
+	int i, len;
+
+	if (size > 0x1fff)
+		goto too_large;
+
+	init_put_bits(&bs, ctx->buffer, size + 1024 + MAX_EXTRADATA_SIZE);
+
+	latmenc_write_frame_header(ctx, extradata, extradata_size, &bs);
+
+	/* PayloadLengthInfo() */
+	for (i = 0; i <= size - 255; i += 255)
+		put_bits(&bs, 8, 255);
+
+	put_bits(&bs, 8, size - i);
+
+	/* The LATM payload is written unaligned */
+
+	/* PayloadMux() */
+	if (size && (data[0] & 0xe1) == 0x81)
+	{
+		// Convert byte-aligned DSE to non-aligned.
+		// Due to the input format encoding we know that
+		// it is naturally byte-aligned in the input stream,
+		// so there are no padding bits to account for.
+		// To avoid having to add padding bits and rearrange
+		// the whole stream we just remove the byte-align flag.
+		// This allows us to remux our FATE AAC samples into latm
+		// files that are still playable with minimal effort.
+		put_bits(&bs, 8, data[0] & 0xfe);
+		avpriv_copy_bits(&bs, data + 1, 8 * size - 8);
+	}
+	else
+		avpriv_copy_bits(&bs, data, 8 * size);
+
+	avpriv_align_put_bits(&bs);
+	flush_put_bits(&bs);
+
+	len = put_bits_count(&bs) >> 3;
+
+	if (len > 0x1fff)
+		goto too_large;
+
+	memcpy(ctx->loas_header, "\x56\xe0\x00", 3);
+	ctx->loas_header[1] |= (len >> 8) & 0x1f;
+	ctx->loas_header[2] |= len & 0xff;
+	ctx->len = len;
+	return 0;
+
+too_large:
+	latmenc_err("LATM packet size larger than maximum size 0x1fff\n");
+	return AVERROR_INVALIDDATA;
+}
+

libeplayer3/external/ffmpeg/src/mpeg4audio.c

@@ -0,0 +1,205 @@
+/*
+ * MPEG-4 Audio common code
+ * Copyright (c) 2008 Baptiste Coudurier <baptiste.coudurier@free.fr>
+ * Copyright (c) 2009 Alex Converse <alex.converse@gmail.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg 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.
+ *
+ * FFmpeg 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 FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include <ffmpeg/get_bits.h>
+#include <ffmpeg/put_bits.h>
+#include <ffmpeg/mpeg4audio.h>
+
+/**
+ * Parse MPEG-4 audio configuration for ALS object type.
+ * @param[in] gb       bit reader context
+ * @param[in] c        MPEG4AudioConfig structure to fill
+ * @return on success 0 is returned, otherwise a value < 0
+ */
+static int parse_config_ALS(GetBitContext *gb, MPEG4AudioConfig *c)
+{
+	if (get_bits_left(gb) < 112)
+		return -1;
+
+	if (get_bits_long(gb, 32) != MKBETAG('A', 'L', 'S', '\0'))
+		return -1;
+
+	// override AudioSpecificConfig channel configuration and sample rate
+	// which are buggy in old ALS conformance files
+	c->sample_rate = get_bits_long(gb, 32);
+
+	// skip number of samples
+	skip_bits_long(gb, 32);
+
+	// read number of channels
+	c->chan_config = 0;
+	c->channels    = get_bits(gb, 16) + 1;
+
+	return 0;
+}
+
+/* XXX: make sure to update the copies in the different encoders if you change
+ * this table */
+const int avpriv_mpeg4audio_sample_rates[16] =
+{
+	96000, 88200, 64000, 48000, 44100, 32000,
+	24000, 22050, 16000, 12000, 11025, 8000, 7350
+};
+
+const uint8_t ff_mpeg4audio_channels[8] =
+{
+	0, 1, 2, 3, 4, 5, 6, 8
+};
+
+static inline int get_object_type(GetBitContext *gb)
+{
+	int object_type = get_bits(gb, 5);
+	if (object_type == AOT_ESCAPE)
+		object_type = 32 + get_bits(gb, 6);
+	return object_type;
+}
+
+static inline int get_sample_rate(GetBitContext *gb, int *index)
+{
+	*index = get_bits(gb, 4);
+	return *index == 0x0f ? (int)get_bits(gb, 24) : avpriv_mpeg4audio_sample_rates[*index];
+}
+
+int avpriv_mpeg4audio_get_config(MPEG4AudioConfig *c, const uint8_t *buf,
+                                 int bit_size, int sync_extension)
+{
+	GetBitContext gb;
+	int specific_config_bitindex, ret;
+
+	if (bit_size <= 0)
+		return AVERROR_INVALIDDATA;
+
+	ret = init_get_bits(&gb, buf, bit_size);
+	if (ret < 0)
+		return ret;
+
+	c->object_type = get_object_type(&gb);
+	c->sample_rate = get_sample_rate(&gb, &c->sampling_index);
+	c->chan_config = get_bits(&gb, 4);
+	if (c->chan_config < (int)FF_ARRAY_ELEMS(ff_mpeg4audio_channels))
+		c->channels = ff_mpeg4audio_channels[c->chan_config];
+	c->sbr = -1;
+	c->ps  = -1;
+	if (c->object_type == AOT_SBR || (c->object_type == AOT_PS &&
+	    // check for W6132 Annex YYYY draft MP3onMP4
+	    !(show_bits(&gb, 3) & 0x03 && !(show_bits(&gb, 9) & 0x3F))))
+	{
+		if (c->object_type == AOT_PS)
+			c->ps = 1;
+		c->ext_object_type = AOT_SBR;
+		c->sbr = 1;
+		c->ext_sample_rate = get_sample_rate(&gb, &c->ext_sampling_index);
+		c->object_type = get_object_type(&gb);
+		if (c->object_type == AOT_ER_BSAC)
+			c->ext_chan_config = get_bits(&gb, 4);
+	}
+	else
+	{
+		c->ext_object_type = AOT_NULL;
+		c->ext_sample_rate = 0;
+	}
+	specific_config_bitindex = get_bits_count(&gb);
+
+	if (c->object_type == AOT_ALS)
+	{
+		skip_bits(&gb, 5);
+		if (show_bits_long(&gb, 24) != MKBETAG('\0', 'A', 'L', 'S'))
+			skip_bits_long(&gb, 24);
+
+		specific_config_bitindex = get_bits_count(&gb);
+
+		if (parse_config_ALS(&gb, c))
+			return -1;
+	}
+
+	if (c->ext_object_type != AOT_SBR && sync_extension)
+	{
+		while (get_bits_left(&gb) > 15)
+		{
+			if (show_bits(&gb, 11) == 0x2b7)   // sync extension
+			{
+				get_bits(&gb, 11);
+				c->ext_object_type = get_object_type(&gb);
+				if (c->ext_object_type == AOT_SBR && (c->sbr = get_bits1(&gb)) == 1)
+				{
+					c->ext_sample_rate = get_sample_rate(&gb, &c->ext_sampling_index);
+					if (c->ext_sample_rate == c->sample_rate)
+						c->sbr = -1;
+				}
+				if (get_bits_left(&gb) > 11 && get_bits(&gb, 11) == 0x548)
+					c->ps = get_bits1(&gb);
+				break;
+			}
+			else
+				get_bits1(&gb); // skip 1 bit
+		}
+	}
+
+	//PS requires SBR
+	if (!c->sbr)
+		c->ps = 0;
+	//Limit implicit PS to the HE-AACv2 Profile
+	if ((c->ps == -1 && c->object_type != AOT_AAC_LC) || c->channels & ~0x01)
+		c->ps = 0;
+
+	return specific_config_bitindex;
+}
+
+static av_always_inline unsigned int copy_bits(PutBitContext *pb,
+        GetBitContext *gb,
+        int bits)
+{
+	unsigned int el = get_bits(gb, bits);
+	put_bits(pb, bits, el);
+	return el;
+}
+
+int avpriv_copy_pce_data(PutBitContext *pb, GetBitContext *gb)
+{
+	int five_bit_ch, four_bit_ch, comment_size, bits;
+	int offset = put_bits_count(pb);
+
+	copy_bits(pb, gb, 10);                  //Tag, Object Type, Frequency
+	five_bit_ch  = copy_bits(pb, gb, 4);    //Front
+	five_bit_ch += copy_bits(pb, gb, 4);    //Side
+	five_bit_ch += copy_bits(pb, gb, 4);    //Back
+	four_bit_ch  = copy_bits(pb, gb, 2);    //LFE
+	four_bit_ch += copy_bits(pb, gb, 3);    //Data
+	five_bit_ch += copy_bits(pb, gb, 4);    //Coupling
+	if (copy_bits(pb, gb, 1))               //Mono Mixdown
+		copy_bits(pb, gb, 4);
+	if (copy_bits(pb, gb, 1))               //Stereo Mixdown
+		copy_bits(pb, gb, 4);
+	if (copy_bits(pb, gb, 1))               //Matrix Mixdown
+		copy_bits(pb, gb, 3);
+	for (bits = five_bit_ch * 5 + four_bit_ch * 4; bits > 16; bits -= 16)
+		copy_bits(pb, gb, 16);
+	if (bits)
+		copy_bits(pb, gb, bits);
+	avpriv_align_put_bits(pb);
+	align_get_bits(gb);
+	comment_size = copy_bits(pb, gb, 8);
+	for (; comment_size > 0; comment_size--)
+		copy_bits(pb, gb, 8);
+
+	return put_bits_count(pb) - offset;
+}