mirror of
https://github.com/hrydgard/ppsspp.git
synced 2026-11-02 20:42:29 +01:00
The ffmpeg submodule held prebuilt libraries for every platform next to the sources (about 1.5 GB checked out), and nothing uses those any more. Only the needed sources are copied in, from the submodule's commit, along with every header and template they include. The path is new, so checking out commits on either side of this works. It's also cut down to what PPSSPP uses: the H.264, AAC, MP3, ATRAC3 and ATRAC3+ decoders, the H.264 parser, the demuxers as before, and the FFV1 encoder, AVI muxer and file protocol for AVI dumping. config.h takes its CONFIG_ values from FFmpeg 3.0.2's own configure run with just those enabled. The raw H.264 demuxer stays even though nothing opens raw H.264: probing PSMF data uses it to identify the video stream's codec, which otherwise comes out as none. On top of that: - The files nothing links (configure would build them, but they're never pulled out of the static library, going by the GCC link map) are left out: mostly libavutil's crypto and hashing and the alternative DCT/IDCT/FFT versions. - libswresample isn't built, now that nothing uses it. - The H.264 decoder only does 8-bit video, which is all the PSP has. The SPS parser rejects any other bit depth, and the DSP files no longer build their 9- to 16-bit variants, which the linker kept because the init functions pick one by bit depth at run time. That's 160 files to compile instead of 277, and 391 of FFmpeg's vendored, about 5.8 MB. The README in ext/ffmpeg has what was done, the PPSSPP patches in the copy, how to add a component, and the tradeoffs. Co-Authored-By: Claude Opus 5.5 (1M context) <noreply@anthropic.com>
339 lines
12 KiB
C
339 lines
12 KiB
C
/*
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* RIFF muxing functions
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* Copyright (c) 2000 Fabrice Bellard
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "libavutil/dict.h"
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#include "libavutil/log.h"
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#include "libavutil/mathematics.h"
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#include "libavcodec/avcodec.h"
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#include "libavcodec/bytestream.h"
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#include "avformat.h"
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#include "avio_internal.h"
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#include "riff.h"
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int64_t ff_start_tag(AVIOContext *pb, const char *tag)
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{
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ffio_wfourcc(pb, tag);
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avio_wl32(pb, -1);
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return avio_tell(pb);
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}
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void ff_end_tag(AVIOContext *pb, int64_t start)
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{
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int64_t pos;
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av_assert0((start&1) == 0);
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pos = avio_tell(pb);
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if (pos & 1)
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avio_w8(pb, 0);
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avio_seek(pb, start - 4, SEEK_SET);
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avio_wl32(pb, (uint32_t)(pos - start));
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avio_seek(pb, FFALIGN(pos, 2), SEEK_SET);
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}
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/* WAVEFORMATEX header */
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/* returns the size or -1 on error */
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int ff_put_wav_header(AVIOContext *pb, AVCodecContext *enc, int flags)
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{
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int bps, blkalign, bytespersec, frame_size;
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int hdrsize;
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int64_t hdrstart = avio_tell(pb);
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int waveformatextensible;
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uint8_t temp[256];
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uint8_t *riff_extradata = temp;
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uint8_t *riff_extradata_start = temp;
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if (!enc->codec_tag || enc->codec_tag > 0xffff)
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return -1;
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/* We use the known constant frame size for the codec if known, otherwise
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* fall back on using AVCodecContext.frame_size, which is not as reliable
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* for indicating packet duration. */
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frame_size = av_get_audio_frame_duration(enc, enc->block_align);
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waveformatextensible = (enc->channels > 2 && enc->channel_layout) ||
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enc->sample_rate > 48000 ||
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enc->codec_id == AV_CODEC_ID_EAC3 ||
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av_get_bits_per_sample(enc->codec_id) > 16;
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if (waveformatextensible)
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avio_wl16(pb, 0xfffe);
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else
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avio_wl16(pb, enc->codec_tag);
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avio_wl16(pb, enc->channels);
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avio_wl32(pb, enc->sample_rate);
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if (enc->codec_id == AV_CODEC_ID_ATRAC3 ||
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enc->codec_id == AV_CODEC_ID_G723_1 ||
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enc->codec_id == AV_CODEC_ID_MP2 ||
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enc->codec_id == AV_CODEC_ID_MP3 ||
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enc->codec_id == AV_CODEC_ID_GSM_MS) {
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bps = 0;
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} else {
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if (!(bps = av_get_bits_per_sample(enc->codec_id))) {
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if (enc->bits_per_coded_sample)
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bps = enc->bits_per_coded_sample;
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else
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bps = 16; // default to 16
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}
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}
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if (bps != enc->bits_per_coded_sample && enc->bits_per_coded_sample) {
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av_log(enc, AV_LOG_WARNING,
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"requested bits_per_coded_sample (%d) "
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"and actually stored (%d) differ\n",
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enc->bits_per_coded_sample, bps);
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}
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if (enc->codec_id == AV_CODEC_ID_MP2) {
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blkalign = (144 * enc->bit_rate - 1)/enc->sample_rate + 1;
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} else if (enc->codec_id == AV_CODEC_ID_MP3) {
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blkalign = 576 * (enc->sample_rate <= (24000 + 32000)/2 ? 1 : 2);
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} else if (enc->codec_id == AV_CODEC_ID_AC3) {
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blkalign = 3840; /* maximum bytes per frame */
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} else if (enc->codec_id == AV_CODEC_ID_AAC) {
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blkalign = 768 * enc->channels; /* maximum bytes per frame */
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} else if (enc->codec_id == AV_CODEC_ID_G723_1) {
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blkalign = 24;
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} else if (enc->block_align != 0) { /* specified by the codec */
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blkalign = enc->block_align;
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} else
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blkalign = bps * enc->channels / av_gcd(8, bps);
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if (enc->codec_id == AV_CODEC_ID_PCM_U8 ||
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enc->codec_id == AV_CODEC_ID_PCM_S24LE ||
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enc->codec_id == AV_CODEC_ID_PCM_S32LE ||
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enc->codec_id == AV_CODEC_ID_PCM_F32LE ||
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enc->codec_id == AV_CODEC_ID_PCM_F64LE ||
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enc->codec_id == AV_CODEC_ID_PCM_S16LE) {
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bytespersec = enc->sample_rate * blkalign;
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} else if (enc->codec_id == AV_CODEC_ID_G723_1) {
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bytespersec = 800;
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} else {
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bytespersec = enc->bit_rate / 8;
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}
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avio_wl32(pb, bytespersec); /* bytes per second */
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avio_wl16(pb, blkalign); /* block align */
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avio_wl16(pb, bps); /* bits per sample */
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if (enc->codec_id == AV_CODEC_ID_MP3) {
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bytestream_put_le16(&riff_extradata, 1); /* wID */
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bytestream_put_le32(&riff_extradata, 2); /* fdwFlags */
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bytestream_put_le16(&riff_extradata, 1152); /* nBlockSize */
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bytestream_put_le16(&riff_extradata, 1); /* nFramesPerBlock */
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bytestream_put_le16(&riff_extradata, 1393); /* nCodecDelay */
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} else if (enc->codec_id == AV_CODEC_ID_MP2) {
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/* fwHeadLayer */
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bytestream_put_le16(&riff_extradata, 2);
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/* dwHeadBitrate */
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bytestream_put_le32(&riff_extradata, enc->bit_rate);
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/* fwHeadMode */
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bytestream_put_le16(&riff_extradata, enc->channels == 2 ? 1 : 8);
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/* fwHeadModeExt */
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bytestream_put_le16(&riff_extradata, 0);
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/* wHeadEmphasis */
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bytestream_put_le16(&riff_extradata, 1);
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/* fwHeadFlags */
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bytestream_put_le16(&riff_extradata, 16);
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/* dwPTSLow */
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bytestream_put_le32(&riff_extradata, 0);
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/* dwPTSHigh */
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bytestream_put_le32(&riff_extradata, 0);
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} else if (enc->codec_id == AV_CODEC_ID_G723_1) {
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bytestream_put_le32(&riff_extradata, 0x9ace0002); /* extradata needed for msacm g723.1 codec */
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bytestream_put_le32(&riff_extradata, 0xaea2f732);
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bytestream_put_le16(&riff_extradata, 0xacde);
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} else if (enc->codec_id == AV_CODEC_ID_GSM_MS ||
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enc->codec_id == AV_CODEC_ID_ADPCM_IMA_WAV) {
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/* wSamplesPerBlock */
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bytestream_put_le16(&riff_extradata, frame_size);
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} else if (enc->extradata_size) {
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riff_extradata_start = enc->extradata;
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riff_extradata = enc->extradata + enc->extradata_size;
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}
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/* write WAVEFORMATEXTENSIBLE extensions */
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if (waveformatextensible) {
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int write_channel_mask = !(flags & FF_PUT_WAV_HEADER_SKIP_CHANNELMASK) &&
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(enc->strict_std_compliance < FF_COMPLIANCE_NORMAL ||
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enc->channel_layout < 0x40000);
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/* 22 is WAVEFORMATEXTENSIBLE size */
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avio_wl16(pb, riff_extradata - riff_extradata_start + 22);
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/* ValidBitsPerSample || SamplesPerBlock || Reserved */
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avio_wl16(pb, bps);
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/* dwChannelMask */
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avio_wl32(pb, write_channel_mask ? enc->channel_layout : 0);
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/* GUID + next 3 */
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if (enc->codec_id == AV_CODEC_ID_EAC3) {
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ff_put_guid(pb, ff_get_codec_guid(enc->codec_id, ff_codec_wav_guids));
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} else {
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avio_wl32(pb, enc->codec_tag);
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avio_wl32(pb, 0x00100000);
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avio_wl32(pb, 0xAA000080);
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avio_wl32(pb, 0x719B3800);
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}
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} else if ((flags & FF_PUT_WAV_HEADER_FORCE_WAVEFORMATEX) ||
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enc->codec_tag != 0x0001 /* PCM */ ||
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riff_extradata - riff_extradata_start) {
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/* WAVEFORMATEX */
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avio_wl16(pb, riff_extradata - riff_extradata_start); /* cbSize */
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} /* else PCMWAVEFORMAT */
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avio_write(pb, riff_extradata_start, riff_extradata - riff_extradata_start);
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hdrsize = avio_tell(pb) - hdrstart;
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if (hdrsize & 1) {
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hdrsize++;
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avio_w8(pb, 0);
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}
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return hdrsize;
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}
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/* BITMAPINFOHEADER header */
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void ff_put_bmp_header(AVIOContext *pb, AVCodecContext *enc,
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const AVCodecTag *tags, int for_asf, int ignore_extradata)
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{
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int keep_height = enc->extradata_size >= 9 &&
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!memcmp(enc->extradata + enc->extradata_size - 9, "BottomUp", 9);
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int extradata_size = enc->extradata_size - 9*keep_height;
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/* size */
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avio_wl32(pb, 40 + (ignore_extradata ? 0 :extradata_size));
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avio_wl32(pb, enc->width);
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//We always store RGB TopDown
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avio_wl32(pb, enc->codec_tag || keep_height ? enc->height : -enc->height);
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/* planes */
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avio_wl16(pb, 1);
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/* depth */
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avio_wl16(pb, enc->bits_per_coded_sample ? enc->bits_per_coded_sample : 24);
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/* compression type */
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avio_wl32(pb, enc->codec_tag);
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avio_wl32(pb, (enc->width * enc->height * (enc->bits_per_coded_sample ? enc->bits_per_coded_sample : 24)+7) / 8);
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avio_wl32(pb, 0);
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avio_wl32(pb, 0);
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avio_wl32(pb, 0);
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avio_wl32(pb, 0);
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if (!ignore_extradata) {
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avio_write(pb, enc->extradata, extradata_size);
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if (!for_asf && extradata_size & 1)
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avio_w8(pb, 0);
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}
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}
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void ff_parse_specific_params(AVStream *st, int *au_rate,
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int *au_ssize, int *au_scale)
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{
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AVCodecContext *codec = st->codec;
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int gcd;
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int audio_frame_size;
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/* We use the known constant frame size for the codec if known, otherwise
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* fall back on using AVCodecContext.frame_size, which is not as reliable
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* for indicating packet duration. */
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audio_frame_size = av_get_audio_frame_duration(codec, 0);
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if (!audio_frame_size)
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audio_frame_size = codec->frame_size;
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*au_ssize = codec->block_align;
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if (audio_frame_size && codec->sample_rate) {
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*au_scale = audio_frame_size;
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*au_rate = codec->sample_rate;
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} else if (codec->codec_type == AVMEDIA_TYPE_VIDEO ||
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codec->codec_type == AVMEDIA_TYPE_DATA ||
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codec->codec_type == AVMEDIA_TYPE_SUBTITLE) {
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*au_scale = st->time_base.num;
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*au_rate = st->time_base.den;
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} else {
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*au_scale = codec->block_align ? codec->block_align * 8 : 8;
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*au_rate = codec->bit_rate ? codec->bit_rate :
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8 * codec->sample_rate;
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}
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gcd = av_gcd(*au_scale, *au_rate);
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*au_scale /= gcd;
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*au_rate /= gcd;
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}
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void ff_riff_write_info_tag(AVIOContext *pb, const char *tag, const char *str)
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{
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size_t len = strlen(str);
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if (len > 0 && len < UINT32_MAX) {
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len++;
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ffio_wfourcc(pb, tag);
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avio_wl32(pb, len);
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avio_put_str(pb, str);
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if (len & 1)
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avio_w8(pb, 0);
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}
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}
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static const char riff_tags[][5] = {
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"IARL", "IART", "ICMS", "ICMT", "ICOP", "ICRD", "ICRP", "IDIM", "IDPI",
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"IENG", "IGNR", "IKEY", "ILGT", "ILNG", "IMED", "INAM", "IPLT", "IPRD",
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"IPRT", "ITRK", "ISBJ", "ISFT", "ISHP", "ISMP", "ISRC", "ISRF", "ITCH",
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{ 0 }
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};
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static int riff_has_valid_tags(AVFormatContext *s)
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{
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int i;
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for (i = 0; *riff_tags[i]; i++)
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if (av_dict_get(s->metadata, riff_tags[i], NULL, AV_DICT_MATCH_CASE))
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return 1;
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return 0;
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}
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void ff_riff_write_info(AVFormatContext *s)
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{
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AVIOContext *pb = s->pb;
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int i;
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int64_t list_pos;
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AVDictionaryEntry *t = NULL;
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ff_metadata_conv(&s->metadata, ff_riff_info_conv, NULL);
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/* writing empty LIST is not nice and may cause problems */
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if (!riff_has_valid_tags(s))
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return;
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list_pos = ff_start_tag(pb, "LIST");
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ffio_wfourcc(pb, "INFO");
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for (i = 0; *riff_tags[i]; i++)
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if ((t = av_dict_get(s->metadata, riff_tags[i],
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NULL, AV_DICT_MATCH_CASE)))
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ff_riff_write_info_tag(s->pb, t->key, t->value);
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ff_end_tag(pb, list_pos);
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}
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void ff_put_guid(AVIOContext *s, const ff_asf_guid *g)
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{
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av_assert0(sizeof(*g) == 16);
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avio_write(s, *g, sizeof(*g));
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}
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const ff_asf_guid *ff_get_codec_guid(enum AVCodecID id, const AVCodecGuid *av_guid)
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{
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int i;
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for (i = 0; av_guid[i].id != AV_CODEC_ID_NONE; i++) {
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if (id == av_guid[i].id)
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return &(av_guid[i].guid);
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}
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return NULL;
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}
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