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-rw-r--r--source/blender/src/hddaudio.c588
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diff --git a/source/blender/src/hddaudio.c b/source/blender/src/hddaudio.c
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index 7175f140b09..00000000000
--- a/source/blender/src/hddaudio.c
+++ /dev/null
@@ -1,588 +0,0 @@
-/**
- * $Id$
- *
- * ***** BEGIN GPL LICENSE BLOCK *****
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
- *
- * This program 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 General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software Foundation,
- * Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
- *
- * Contributor(s): Peter Schlaile <peter [at] schlaile [dot] de> 2005
- *
- * ***** END GPL LICENSE BLOCK *****
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <math.h>
-#include <string.h>
-#include <fcntl.h>
-
-#ifdef HAVE_CONFIG_H
-#include <config.h>
-#endif
-
-#ifdef WITH_FFMPEG
-#include <libavformat/avformat.h>
-#include <libavcodec/avcodec.h>
-#include <libavutil/rational.h>
-#if LIBAVFORMAT_VERSION_INT < (49 << 16)
-#define FFMPEG_OLD_FRAME_RATE 1
-#else
-#define FFMPEG_CODEC_IS_POINTER 1
-#endif
-#endif
-
-#include "MEM_guardedalloc.h"
-
-#include "BIF_editsound.h"
-
-#include "blendef.h"
-
-#ifdef WITH_FFMPEG
-extern void do_init_ffmpeg();
-#endif
-
-struct hdaudio {
- int sample_rate;
- int channels;
- int audioStream;
-
-#ifdef WITH_FFMPEG
- char * filename;
- AVCodec *pCodec;
- AVFormatContext *pFormatCtx;
- AVCodecContext *pCodecCtx;
- int frame_position;
- int frame_duration;
- int frame_alloc_duration;
- int decode_pos;
- int frame_size;
- short * decode_cache;
- short * decode_cache_zero;
- short * resample_cache;
- int decode_cache_size;
- int target_channels;
- int target_rate;
- int resample_samples_written;
- int resample_samples_in;
- ReSampleContext *resampler;
-#else
-
-
-#endif
-};
-
-#ifdef WITH_FFMPEG
-#ifdef FFMPEG_CODEC_IS_POINTER
-static AVCodecContext* get_codec_from_stream(AVStream* stream)
-{
- return stream->codec;
-}
-#else
-static AVCodecContext* get_codec_from_stream(AVStream* stream)
-{
- return &stream->codec;
-}
-#endif
-#endif
-
-struct hdaudio * sound_open_hdaudio(char * filename)
-{
-#ifdef WITH_FFMPEG
- struct hdaudio * rval;
- int i, audioStream;
-
- AVCodec *pCodec;
- AVFormatContext *pFormatCtx;
- AVCodecContext *pCodecCtx;
-
- do_init_ffmpeg();
-
- if(av_open_input_file(&pFormatCtx, filename, NULL, 0, NULL)!=0) {
- return 0;
- }
-
- if(av_find_stream_info(pFormatCtx)<0) {
- av_close_input_file(pFormatCtx);
- return 0;
- }
-
- dump_format(pFormatCtx, 0, filename, 0);
-
-
- /* Find the first audio stream */
- audioStream=-1;
- for(i=0; i<pFormatCtx->nb_streams; i++)
- if(get_codec_from_stream(pFormatCtx->streams[i])
- ->codec_type == CODEC_TYPE_AUDIO)
- {
- audioStream=i;
- break;
- }
-
- if(audioStream == -1) {
- av_close_input_file(pFormatCtx);
- return 0;
- }
-
- pCodecCtx = get_codec_from_stream(pFormatCtx->streams[audioStream]);
-
- /* Find the decoder for the audio stream */
- pCodec = avcodec_find_decoder(pCodecCtx->codec_id);
- if(pCodec == NULL) {
- av_close_input_file(pFormatCtx);
- return 0;
- }
-
- if(avcodec_open(pCodecCtx, pCodec)<0) {
- av_close_input_file(pFormatCtx);
- return 0;
- }
-
- if (pCodecCtx->channels > 2) {
- fprintf(stderr, "Sorry, audio file has too many channels."
- " Will fix in future, but resampler doesn't support "
- "this.\n");
- avcodec_close(pCodecCtx);
- av_close_input_file(pFormatCtx);
- return 0;
- }
-
- rval = (struct hdaudio *)MEM_mallocN(sizeof(struct hdaudio),
- "hdaudio struct");
-
- rval->filename = strdup(filename);
- rval->sample_rate = pCodecCtx->sample_rate;
- rval->channels = pCodecCtx->channels;
-
- rval->pFormatCtx = pFormatCtx;
- rval->pCodecCtx = pCodecCtx;
- rval->pCodec = pCodec;
- rval->audioStream = audioStream;
- rval->frame_position = -10;
-
- rval->frame_duration = AV_TIME_BASE / 10;
- rval->frame_alloc_duration = AV_TIME_BASE;
- rval->decode_cache_size =
- (long long) rval->sample_rate * rval->channels
- * rval->frame_alloc_duration / AV_TIME_BASE
- * 2;
-
- rval->decode_cache = (short*) MEM_mallocN(
- rval->decode_cache_size * sizeof(short)
- + AVCODEC_MAX_AUDIO_FRAME_SIZE,
- "hdaudio decode cache");
- rval->decode_cache_zero = rval->decode_cache;
- rval->decode_pos = 0;
- rval->target_channels = -1;
- rval->target_rate = -1;
- rval->resampler = 0;
- rval->resample_cache = 0;
- rval->resample_samples_written = 0;
- rval->resample_samples_in = 0;
- return rval;
-#else
- return 0;
-#endif
-}
-
-struct hdaudio * sound_copy_hdaudio(struct hdaudio * c)
-{
-#ifdef WITH_FFMPEG
- return sound_open_hdaudio(c->filename);
-#else
- return 0;
-#endif
-}
-
-long sound_hdaudio_get_duration(struct hdaudio * hdaudio, double frame_rate)
-{
-#ifdef WITH_FFMPEG
- return hdaudio->pFormatCtx->duration * frame_rate / AV_TIME_BASE;
-#else
- return 0;
-#endif
-}
-
-#ifdef WITH_FFMPEG
-static void sound_hdaudio_extract_small_block(
- struct hdaudio * hdaudio,
- short * target_buffer,
- int sample_position /* units of target_rate */,
- int target_rate,
- int target_channels,
- int nb_samples /* in target */)
-{
- AVPacket packet;
- int frame_position;
- int frame_size = (long long) target_rate
- * hdaudio->frame_duration / AV_TIME_BASE;
- int in_frame_size = (long long) hdaudio->sample_rate
- * hdaudio->frame_duration / AV_TIME_BASE;
- int rate_conversion =
- (target_rate != hdaudio->sample_rate)
- || (target_channels != hdaudio->channels);
- int sample_ofs = target_channels * (sample_position % frame_size);
-
- frame_position = sample_position / frame_size;
-
- if (hdaudio == 0) return;
-
- if (rate_conversion) {
- if (hdaudio->resampler &&
- (hdaudio->target_rate != target_rate
- || hdaudio->target_channels != target_channels)) {
- audio_resample_close(hdaudio->resampler);
- hdaudio->resampler = 0;
- }
- if (!hdaudio->resampler) {
- hdaudio->resampler = audio_resample_init(
- target_channels, hdaudio->channels,
- target_rate, hdaudio->sample_rate);
- hdaudio->target_rate = target_rate;
- hdaudio->target_channels = target_channels;
- if (hdaudio->resample_cache) {
- MEM_freeN(hdaudio->resample_cache);
- }
-
-
- hdaudio->resample_cache = (short*) MEM_mallocN(
- (long long)
- hdaudio->target_channels
- * frame_size * 2
- * sizeof(short),
- "hdaudio resample cache");
- if (frame_position == hdaudio->frame_position) {
- hdaudio->resample_samples_in =
- in_frame_size * 7 / 4;
- hdaudio->resample_samples_written
- = audio_resample(
- hdaudio->resampler,
- hdaudio->resample_cache,
- hdaudio->decode_cache_zero,
- in_frame_size * 7 / 4);
- }
- }
- }
-
- if (frame_position == hdaudio->frame_position + 1
- && in_frame_size * hdaudio->channels <= hdaudio->decode_pos) {
- int bl_size = in_frame_size * hdaudio->channels;
- int decode_pos = hdaudio->decode_pos;
-
- hdaudio->frame_position = frame_position;
-
- memmove(hdaudio->decode_cache,
- hdaudio->decode_cache + bl_size,
- (decode_pos - bl_size) * sizeof(short));
-
- decode_pos -= bl_size;
-
- if (decode_pos < hdaudio->decode_cache_size) {
- memset(hdaudio->decode_cache + decode_pos, 0,
- (hdaudio->decode_cache_size - decode_pos)
- * sizeof(short));
-
- while(av_read_frame(
- hdaudio->pFormatCtx, &packet) >= 0) {
- int data_size;
- int len;
- uint8_t *audio_pkt_data;
- int audio_pkt_size;
-
- if(packet.stream_index
- != hdaudio->audioStream) {
- av_free_packet(&packet);
- continue;
- }
-
- audio_pkt_data = packet.data;
- audio_pkt_size = packet.size;
-
- while (audio_pkt_size > 0) {
- data_size=AVCODEC_MAX_AUDIO_FRAME_SIZE;
- len = avcodec_decode_audio2(
- hdaudio->pCodecCtx,
- hdaudio->decode_cache
- + decode_pos,
- &data_size,
- audio_pkt_data,
- audio_pkt_size);
- if (len <= 0) {
- audio_pkt_size = 0;
- break;
- }
-
- audio_pkt_size -= len;
- audio_pkt_data += len;
-
- if (data_size <= 0) {
- continue;
- }
-
- decode_pos += data_size / sizeof(short);
- if (decode_pos + data_size
- / sizeof(short)
- > hdaudio->decode_cache_size) {
- break;
- }
- }
- av_free_packet(&packet);
-
- if (decode_pos + data_size / sizeof(short)
- > hdaudio->decode_cache_size) {
- break;
- }
- }
- }
-
- if (rate_conversion) {
- int written = hdaudio->resample_samples_written
- * target_channels;
- int ofs = target_channels * frame_size;
- int recycle = written - ofs;
- int next_in = in_frame_size
- + (3.0/4.0
- - (double) recycle / (double)
- (frame_size * target_channels)
- ) * in_frame_size;
-
- memmove(hdaudio->resample_cache,
- hdaudio->resample_cache + ofs,
- recycle * sizeof(short));
-
- hdaudio->resample_samples_written
- = audio_resample(
- hdaudio->resampler,
- hdaudio->resample_cache + recycle,
- hdaudio->decode_cache_zero
- + hdaudio->resample_samples_in
- * hdaudio->channels
- - bl_size,
- next_in)
- + recycle / target_channels;
-
- hdaudio->resample_samples_in = next_in;
- }
-
- hdaudio->decode_pos = decode_pos;
- }
-
- if (frame_position != hdaudio->frame_position) {
- long decode_pos = 0;
- long long st_time = hdaudio->pFormatCtx
- ->streams[hdaudio->audioStream]->start_time;
- double time_base =
- av_q2d(hdaudio->pFormatCtx
- ->streams[hdaudio->audioStream]->time_base);
- long long pos = (long long) frame_position * AV_TIME_BASE
- * hdaudio->frame_duration / AV_TIME_BASE;
-
- long long seek_pos;
-
- hdaudio->frame_position = frame_position;
-
- if (st_time == AV_NOPTS_VALUE) {
- st_time = 0;
- }
-
- pos += st_time * AV_TIME_BASE * time_base;
-
- /* seek a little bit before the target position,
- (ffmpeg seek algorithm doesn't seem to work always as
- specified...)
- */
-
- seek_pos = pos - (AV_TIME_BASE
- * hdaudio->frame_duration
- / AV_TIME_BASE / 10);
- if (seek_pos < 0) {
- seek_pos = pos;
- }
-
- av_seek_frame(hdaudio->pFormatCtx, -1,
- seek_pos,
- AVSEEK_FLAG_ANY | AVSEEK_FLAG_BACKWARD);
- avcodec_flush_buffers(hdaudio->pCodecCtx);
-
- memset(hdaudio->decode_cache, 0,
- hdaudio->decode_cache_size * sizeof(short));
-
- hdaudio->decode_cache_zero = hdaudio->decode_cache;
-
- while(av_read_frame(hdaudio->pFormatCtx, &packet) >= 0) {
- int data_size;
- int len;
- uint8_t *audio_pkt_data;
- int audio_pkt_size;
-
- if(packet.stream_index != hdaudio->audioStream) {
- av_free_packet(&packet);
- continue;
- }
-
- audio_pkt_data = packet.data;
- audio_pkt_size = packet.size;
-
- if (!hdaudio->decode_cache_zero
- && audio_pkt_size > 0) {
- long long diff;
-
- if (packet.pts == AV_NOPTS_VALUE) {
- fprintf(stderr,
- "hdaudio: audio "
- "pts=NULL audio "
- "distortion!\n");
- diff = 0;
- } else {
- long long pts = packet.pts;
- long long spts = (long long) (
- pos / time_base / AV_TIME_BASE
- + 0.5);
- diff = spts - pts;
- if (diff < 0) {
- fprintf(stderr,
- "hdaudio: "
- "negative seek: "
- "%lld < %lld "
- "(pos=%lld) "
- "audio distortion!!\n",
- spts, pts, pos);
- diff = 0;
- }
- }
-
-
- diff *= hdaudio->sample_rate * time_base;
- diff *= hdaudio->channels;
-
- if (diff > hdaudio->decode_cache_size / 2) {
- fprintf(stderr,
- "hdaudio: audio "
- "diff too large!!\n");
- diff = 0;
- }
-
- hdaudio->decode_cache_zero
- = hdaudio->decode_cache + diff;
- }
-
- while (audio_pkt_size > 0) {
- data_size=AVCODEC_MAX_AUDIO_FRAME_SIZE;
- len = avcodec_decode_audio2(
- hdaudio->pCodecCtx,
- hdaudio->decode_cache
- + decode_pos,
- &data_size,
- audio_pkt_data,
- audio_pkt_size);
- if (len <= 0) {
- audio_pkt_size = 0;
- break;
- }
-
- audio_pkt_size -= len;
- audio_pkt_data += len;
-
- if (data_size <= 0) {
- continue;
- }
-
- decode_pos += data_size / sizeof(short);
- if (decode_pos + data_size
- / sizeof(short)
- > hdaudio->decode_cache_size) {
- break;
- }
- }
-
- av_free_packet(&packet);
-
- if (decode_pos + data_size / sizeof(short)
- > hdaudio->decode_cache_size) {
- break;
- }
- }
- if (rate_conversion) {
- hdaudio->resample_samples_written
- = audio_resample(hdaudio->resampler,
- hdaudio->resample_cache,
- hdaudio->decode_cache_zero,
- in_frame_size * 7 / 4);
- hdaudio->resample_samples_in =
- in_frame_size * 7 / 4;
- }
- hdaudio->decode_pos = decode_pos;
- }
-
- memcpy(target_buffer, (rate_conversion
- ? hdaudio->resample_cache
- : hdaudio->decode_cache_zero) + sample_ofs,
- nb_samples * target_channels * sizeof(short));
-}
-#endif
-
-
-void sound_hdaudio_extract(struct hdaudio * hdaudio,
- short * target_buffer,
- int sample_position /* units of target_rate */,
- int target_rate,
- int target_channels,
- int nb_samples /* in target */)
-{
-#ifdef WITH_FFMPEG
- long long max_samples = (long long) target_rate
- * hdaudio->frame_duration / AV_TIME_BASE / 4;
-
- while (nb_samples > max_samples) {
- sound_hdaudio_extract_small_block(hdaudio, target_buffer,
- sample_position,
- target_rate,
- target_channels,
- max_samples);
- target_buffer += max_samples * target_channels;
- sample_position += max_samples;
- nb_samples -= max_samples;
- }
- if (nb_samples > 0) {
- sound_hdaudio_extract_small_block(hdaudio, target_buffer,
- sample_position,
- target_rate,
- target_channels,
- nb_samples);
- }
-#else
-
-#endif
-}
-
-void sound_close_hdaudio(struct hdaudio * hdaudio)
-{
-#ifdef WITH_FFMPEG
-
- if (hdaudio) {
- avcodec_close(hdaudio->pCodecCtx);
- av_close_input_file(hdaudio->pFormatCtx);
- MEM_freeN (hdaudio->decode_cache);
- if (hdaudio->resample_cache) {
- MEM_freeN(hdaudio->resample_cache);
- }
- free(hdaudio->filename);
- MEM_freeN (hdaudio);
- }
-#else
-
-#endif
-}
-
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