1050 lines
36 KiB
C++
1050 lines
36 KiB
C++
// ZoneMinder Video Storage Implementation
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// Written by Chris Wiggins
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// http://chriswiggins.co.nz
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// Modification by Steve Gilvarry
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//
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// This program is free software; you can redistribute it and/or
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// modify it under the terms of the GNU General Public License
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// as published by the Free Software Foundation; either version 2
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// of the License, or (at your option) any later version.
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//
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// This program 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
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// GNU General Public License for more details.
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//
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// You should have received a copy of the GNU General Public License
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// along with this program; if not, write to the Free Software
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// Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
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//
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#define __STDC_FORMAT_MACROS 1
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#include <inttypes.h>
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#include <stdlib.h>
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#include <string.h>
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#include "zm.h"
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#include "zm_videostore.h"
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extern "C" {
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#include "libavutil/time.h"
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}
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VideoStore::VideoStore(const char *filename_in, const char *format_in,
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AVStream *p_video_in_stream,
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AVStream *p_audio_in_stream, int64_t nStartTime,
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Monitor *monitor) {
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video_in_stream = p_video_in_stream;
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audio_in_stream = p_audio_in_stream;
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#if LIBAVCODEC_VERSION_CHECK(57, 64, 0, 64, 0)
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video_in_ctx = avcodec_alloc_context3(NULL);
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avcodec_parameters_to_context(video_in_ctx,
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video_in_stream->codecpar);
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// zm_dump_codecpar( video_in_stream->codecpar );
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#else
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video_in_ctx = video_in_stream->codec;
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#endif
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// store ins in variables local to class
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filename = filename_in;
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format = format_in;
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Info("Opening video storage stream %s format: %s", filename, format);
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ret = avformat_alloc_output_context2(&oc, NULL, NULL, filename);
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if (ret < 0) {
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Warning(
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"Could not create video storage stream %s as no out ctx"
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" could be assigned based on filename: %s",
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filename, av_make_error_string(ret).c_str());
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} else {
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Debug(4, "Success allocating out format ctx");
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}
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// Couldn't deduce format from filename, trying from format name
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if (!oc) {
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avformat_alloc_output_context2(&oc, NULL, format, filename);
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if (!oc) {
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Fatal(
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"Could not create video storage stream %s as no out ctx"
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" could not be assigned based on filename or format %s",
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filename, format);
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} else {
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Debug(4, "Success alocateing out ctx");
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}
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} // end if ! oc
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AVDictionary *pmetadata = NULL;
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int dsr =
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av_dict_set(&pmetadata, "title", "Zoneminder Security Recording", 0);
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if (dsr < 0) Warning("%s:%d: title set failed", __FILE__, __LINE__);
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oc->metadata = pmetadata;
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out_format = oc->oformat;
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#if LIBAVCODEC_VERSION_CHECK(57, 64, 0, 64, 0)
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// Since we are not re-encoding, all we have to do is copy the parameters
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video_out_ctx = avcodec_alloc_context3(NULL);
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// Copy params from instream to ctx
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ret = avcodec_parameters_to_context(video_out_ctx,
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video_in_stream->codecpar);
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if (ret < 0) {
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Error("Could not initialize ctx parameteres");
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return;
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} else {
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zm_dump_codec(video_out_ctx);
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}
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video_out_stream = avformat_new_stream(oc, NULL);
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if (!video_out_stream) {
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Fatal("Unable to create video out stream\n");
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} else {
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Debug(2, "Success creating video out stream");
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}
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if (!video_out_ctx->codec_tag) {
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video_out_ctx->codec_tag =
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av_codec_get_tag(oc->oformat->codec_tag, video_in_ctx->codec_id);
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Debug(2, "No codec_tag, setting to %d", video_out_ctx->codec_tag);
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}
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// Now copy them to the out stream
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ret = avcodec_parameters_from_context(video_out_stream->codecpar,
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video_out_ctx);
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if (ret < 0) {
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Error("Could not initialize stream parameteres");
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return;
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} else {
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Debug(2, "Success setting parameters");
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}
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zm_dump_codecpar(video_out_stream->codecpar);
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#else
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video_out_stream =
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avformat_new_stream(oc,(const AVCodec *)(video_in_ctx->codec));
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if (!video_out_stream) {
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Fatal("Unable to create video out stream\n");
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} else {
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Debug(2, "Success creating video out stream");
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}
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video_out_ctx = video_out_stream->codec;
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ret = avcodec_copy_context(video_out_ctx, video_in_ctx);
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if (ret < 0) {
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Fatal("Unable to copy in video ctx to out video ctx %s\n",
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av_make_error_string(ret).c_str());
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} else {
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Debug(3, "Success copying ctx");
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}
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if (!video_out_ctx->codec_tag) {
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Debug(2, "No codec_tag");
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if (!oc->oformat->codec_tag ||
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av_codec_get_id(oc->oformat->codec_tag,
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video_in_ctx->codec_tag) ==
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video_out_ctx->codec_id ||
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av_codec_get_tag(oc->oformat->codec_tag,
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video_in_ctx->codec_id) <= 0) {
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Warning("Setting codec tag");
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video_out_ctx->codec_tag = video_in_ctx->codec_tag;
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}
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}
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#endif
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// Just copy them from the in, no reason to choose different
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video_out_ctx->time_base = video_in_ctx->time_base;
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video_out_stream->time_base = video_in_stream->time_base;
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Debug(3,
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"Time bases: VIDEO in stream (%d/%d) in codec: (%d/%d) out "
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"stream: (%d/%d) out codec (%d/%d)",
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video_in_stream->time_base.num, video_in_stream->time_base.den,
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video_in_ctx->time_base.num, video_in_ctx->time_base.den,
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video_out_stream->time_base.num, video_out_stream->time_base.den,
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video_out_ctx->time_base.num,
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video_out_ctx->time_base.den);
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if (oc->oformat->flags & AVFMT_GLOBALHEADER) {
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video_out_ctx->flags |= CODEC_FLAG_GLOBAL_HEADER;
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}
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Monitor::Orientation orientation = monitor->getOrientation();
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Debug(3, "Have orientation");
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if (orientation) {
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if (orientation == Monitor::ROTATE_0) {
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} else if (orientation == Monitor::ROTATE_90) {
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dsr = av_dict_set(&video_out_stream->metadata, "rotate", "90", 0);
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if (dsr < 0) Warning("%s:%d: title set failed", __FILE__, __LINE__);
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} else if (orientation == Monitor::ROTATE_180) {
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dsr = av_dict_set(&video_out_stream->metadata, "rotate", "180", 0);
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if (dsr < 0) Warning("%s:%d: title set failed", __FILE__, __LINE__);
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} else if (orientation == Monitor::ROTATE_270) {
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dsr = av_dict_set(&video_out_stream->metadata, "rotate", "270", 0);
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if (dsr < 0) Warning("%s:%d: title set failed", __FILE__, __LINE__);
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} else {
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Warning("Unsupported Orientation(%d)", orientation);
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}
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}
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converted_in_samples = NULL;
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audio_out_codec = NULL;
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audio_in_ctx = NULL;
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audio_out_stream = NULL;
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in_frame = NULL;
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out_frame = NULL;
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#ifdef HAVE_LIBAVRESAMPLE
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resample_ctx = NULL;
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#endif
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if (audio_in_stream) {
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Debug(3, "Have audio stream");
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#if LIBAVCODEC_VERSION_CHECK(57, 64, 0, 64, 0)
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audio_in_ctx = avcodec_alloc_context3(NULL);
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ret = avcodec_parameters_to_context(audio_in_ctx,
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audio_in_stream->codecpar);
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#else
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audio_in_ctx = audio_in_stream->codec;
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#endif
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if (audio_in_ctx->codec_id != AV_CODEC_ID_AAC) {
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static char error_buffer[256];
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avcodec_string(error_buffer, sizeof(error_buffer), audio_in_ctx,
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0);
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Debug(2, "Got something other than AAC (%s)", error_buffer);
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if (!setup_resampler()) {
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return;
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}
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} else {
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Debug(3, "Got AAC");
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audio_out_stream =
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avformat_new_stream(oc, (const AVCodec *)(audio_in_ctx->codec));
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if (!audio_out_stream) {
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Error("Unable to create audio out stream\n");
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audio_out_stream = NULL;
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} else {
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Debug(2, "setting parameters");
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#if LIBAVCODEC_VERSION_CHECK(57, 64, 0, 64, 0)
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audio_out_ctx = avcodec_alloc_context3(audio_out_codec);
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// Copy params from instream to ctx
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ret = avcodec_parameters_to_context(audio_out_ctx,
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audio_in_stream->codecpar);
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if (ret < 0) {
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Error("Unable to copy audio params to ctx %s\n",
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av_make_error_string(ret).c_str());
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}
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ret = avcodec_parameters_from_context(audio_out_stream->codecpar,
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audio_out_ctx);
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if (ret < 0) {
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Error("Unable to copy audio params to stream %s\n",
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av_make_error_string(ret).c_str());
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}
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if (!audio_out_ctx->codec_tag) {
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audio_out_ctx->codec_tag = av_codec_get_tag(
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oc->oformat->codec_tag, audio_in_ctx->codec_id);
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Debug(2, "Setting audio codec tag to %d",
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audio_out_ctx->codec_tag);
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}
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#else
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audio_out_ctx = audio_out_stream->codec;
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ret = avcodec_copy_context(audio_out_ctx, audio_in_ctx);
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audio_out_ctx->codec_tag = 0;
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#endif
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if (ret < 0) {
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Error("Unable to copy audio ctx %s\n",
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av_make_error_string(ret).c_str());
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audio_out_stream = NULL;
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} else {
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if (audio_out_ctx->channels > 1) {
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Warning("Audio isn't mono, changing it.");
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audio_out_ctx->channels = 1;
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} else {
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Debug(3, "Audio is mono");
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}
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}
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} // end if audio_out_stream
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} // end if is AAC
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if (audio_out_stream) {
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if (oc->oformat->flags & AVFMT_GLOBALHEADER) {
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audio_out_ctx->flags |= CODEC_FLAG_GLOBAL_HEADER;
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}
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}
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} // end if audio_in_stream
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video_last_pts = 0;
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video_last_dts = 0;
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audio_last_pts = 0;
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audio_last_dts = 0;
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video_next_pts = 0;
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video_next_dts = 0;
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audio_next_pts = 0;
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audio_next_dts = 0;
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} // VideoStore::VideoStore
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bool VideoStore::open() {
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/* open the out file, if needed */
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if (!(out_format->flags & AVFMT_NOFILE)) {
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ret = avio_open2(&oc->pb, filename, AVIO_FLAG_WRITE, NULL, NULL);
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if (ret < 0) {
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Error("Could not open out file '%s': %s\n", filename,
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av_make_error_string(ret).c_str());
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return false;
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}
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}
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// os->ctx_inited = 1;
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// avio_flush(ctx->pb);
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// av_dict_free(&opts);
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zm_dump_stream_format(oc, 0, 0, 1);
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if (audio_out_stream) zm_dump_stream_format(oc, 1, 0, 1);
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AVDictionary *opts = NULL;
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// av_dict_set(&opts, "movflags", "frag_custom+dash+delay_moov", 0);
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// av_dict_set(&opts, "movflags", "frag_custom+dash+delay_moov", 0);
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// av_dict_set(&opts, "movflags",
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// "frag_keyframe+empty_moov+default_base_moof", 0);
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if ((ret = avformat_write_header(oc, &opts)) < 0) {
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// if ((ret = avformat_write_header(oc, &opts)) < 0) {
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Warning("Unable to set movflags to frag_custom+dash+delay_moov");
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/* Write the stream header, if any. */
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ret = avformat_write_header(oc, NULL);
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} else if (av_dict_count(opts) != 0) {
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Warning("some options not set\n");
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}
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if (ret < 0) {
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Error("Error occurred when writing out file header to %s: %s\n",
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filename, av_make_error_string(ret).c_str());
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return false;
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}
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if (opts) av_dict_free(&opts);
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return true;
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}
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VideoStore::~VideoStore() {
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if (audio_out_codec) {
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// The codec queues data. We need to send a flush command and out
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// whatever we get. Failures are not fatal.
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AVPacket pkt;
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av_init_packet(&pkt);
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while (1) {
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#if LIBAVCODEC_VERSION_CHECK(57, 64, 0, 64, 0)
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// Put encoder into flushing mode
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avcodec_send_frame(audio_out_ctx, NULL);
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ret = avcodec_receive_packet(audio_out_ctx, &pkt);
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if (ret < 0) {
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if (AVERROR_EOF != ret) {
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Error("ERror encoding audio while flushing (%d) (%s)", ret,
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av_err2str(ret));
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}
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break;
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}
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#else
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int got_packet = 0;
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ret =
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avcodec_encode_audio2(audio_out_ctx, &pkt, NULL, &got_packet);
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if (ret < 0) {
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Error("ERror encoding audio while flushing (%d) (%s)", ret,
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av_err2str(ret));
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break;
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}
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Debug(1, "Have audio encoder, need to flush it's out");
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if (!got_packet) {
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break;
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}
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#endif
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Debug(2, "writing flushed packet pts(%d) dts(%d) duration(%d)", pkt.pts,
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pkt.dts, pkt.duration);
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pkt.pts = audio_next_pts;
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pkt.dts = audio_next_dts;
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if (pkt.duration > 0)
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pkt.duration =
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av_rescale_q(pkt.duration, audio_out_ctx->time_base,
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audio_out_stream->time_base);
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audio_next_pts += pkt.duration;
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audio_next_dts += pkt.duration;
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Debug(2, "writing flushed packet pts(%d) dts(%d) duration(%d)", pkt.pts,
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pkt.dts, pkt.duration);
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pkt.stream_index = audio_out_stream->index;
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av_interleaved_write_frame(oc, &pkt);
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zm_av_packet_unref(&pkt);
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} // while have buffered frames
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} // end if audio_out_codec
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// Flush Queues
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av_interleaved_write_frame(oc, NULL);
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/* Write the trailer before close */
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if (int rc = av_write_trailer(oc)) {
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Error("Error writing trailer %s", av_err2str(rc));
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} else {
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Debug(3, "Sucess Writing trailer");
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}
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// I wonder if we should be closing the file first.
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// I also wonder if we really need to be doing all the ctx
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// allocation/de-allocation constantly, or whether we can just re-use it.
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// Just do a file open/close/writeheader/etc.
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// What if we were only doing audio recording?
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if (video_out_stream) {
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avcodec_close(video_out_ctx);
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video_out_ctx = NULL;
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Debug(4, "Success freeing video_out_ctx");
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}
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if (audio_out_stream) {
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avcodec_close(audio_out_ctx);
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audio_out_ctx = NULL;
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#ifdef HAVE_LIBAVRESAMPLE
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if (resample_ctx) {
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avresample_close(resample_ctx);
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avresample_free(&resample_ctx);
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}
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if (in_frame) {
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av_frame_free(&in_frame);
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in_frame = NULL;
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}
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if (out_frame) {
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av_frame_free(&out_frame);
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out_frame = NULL;
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}
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if (converted_in_samples) {
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av_free(converted_in_samples);
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converted_in_samples = NULL;
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}
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#endif
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}
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// WHen will be not using a file ?
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if (!(out_format->flags & AVFMT_NOFILE)) {
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/* Close the out file. */
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if (int rc = avio_close(oc->pb)) {
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Error("Error closing avio %s", av_err2str(rc));
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}
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} else {
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Debug(3, "Not closing avio because we are not writing to a file.");
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}
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/* free the stream */
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avformat_free_context(oc);
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}
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bool VideoStore::setup_resampler() {
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#ifdef HAVE_LIBAVRESAMPLE
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static char error_buffer[256];
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#if LIBAVCODEC_VERSION_CHECK(57, 64, 0, 64, 0)
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// Newer ffmpeg wants to keep everything separate... so have to lookup our own
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// decoder, can't reuse the one from the camera.
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AVCodec *audio_in_codec =
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avcodec_find_decoder(audio_in_stream->codecpar->codec_id);
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#else
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AVCodec *audio_in_codec =
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avcodec_find_decoder(audio_in_ctx->codec_id);
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#endif
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ret = avcodec_open2(audio_in_ctx, audio_in_codec, NULL);
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if (ret < 0) {
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Error("Can't open in codec!");
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return false;
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}
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audio_out_codec = avcodec_find_encoder(AV_CODEC_ID_AAC);
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if (!audio_out_codec) {
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Error("Could not find codec for AAC");
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return false;
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}
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Debug(2, "Have audio out codec");
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// audio_out_ctx = audio_out_stream->codec;
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audio_out_ctx = avcodec_alloc_context3(audio_out_codec);
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if (!audio_out_ctx) {
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Error("could not allocate codec ctx for AAC\n");
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audio_out_stream = NULL;
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return false;
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}
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Debug(2, "Have audio_out_ctx");
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/* put sample parameters */
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audio_out_ctx->bit_rate = audio_in_ctx->bit_rate;
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audio_out_ctx->sample_rate = audio_in_ctx->sample_rate;
|
|
audio_out_ctx->channels = audio_in_ctx->channels;
|
|
audio_out_ctx->channel_layout = audio_in_ctx->channel_layout;
|
|
audio_out_ctx->sample_fmt = audio_in_ctx->sample_fmt;
|
|
audio_out_ctx->refcounted_frames = 1;
|
|
|
|
if (audio_out_codec->supported_samplerates) {
|
|
int found = 0;
|
|
for (unsigned int i = 0; audio_out_codec->supported_samplerates[i];
|
|
i++) {
|
|
if (audio_out_ctx->sample_rate ==
|
|
audio_out_codec->supported_samplerates[i]) {
|
|
found = 1;
|
|
break;
|
|
}
|
|
}
|
|
if (found) {
|
|
Debug(3, "Sample rate is good");
|
|
} else {
|
|
audio_out_ctx->sample_rate =
|
|
audio_out_codec->supported_samplerates[0];
|
|
Debug(1, "Sampel rate is no good, setting to (%d)",
|
|
audio_out_codec->supported_samplerates[0]);
|
|
}
|
|
}
|
|
|
|
/* check that the encoder supports s16 pcm in */
|
|
if (!check_sample_fmt(audio_out_codec, audio_out_ctx->sample_fmt)) {
|
|
Debug(3, "Encoder does not support sample format %s, setting to FLTP",
|
|
av_get_sample_fmt_name(audio_out_ctx->sample_fmt));
|
|
audio_out_ctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
|
|
}
|
|
|
|
audio_out_ctx->time_base =
|
|
(AVRational){1, audio_out_ctx->sample_rate};
|
|
|
|
// Now copy them to the out stream
|
|
audio_out_stream = avformat_new_stream(oc, audio_out_codec);
|
|
|
|
#if LIBAVCODEC_VERSION_CHECK(57, 64, 0, 64, 0)
|
|
ret = avcodec_parameters_from_context(audio_out_stream->codecpar,
|
|
audio_out_ctx);
|
|
if (ret < 0) {
|
|
Error("Could not initialize stream parameteres");
|
|
return false;
|
|
}
|
|
#endif
|
|
|
|
AVDictionary *opts = NULL;
|
|
av_dict_set(&opts, "strict", "experimental", 0);
|
|
ret = avcodec_open2(audio_out_ctx, audio_out_codec, &opts);
|
|
av_dict_free(&opts);
|
|
if (ret < 0) {
|
|
av_strerror(ret, error_buffer, sizeof(error_buffer));
|
|
Fatal("could not open codec (%d) (%s)\n", ret, error_buffer);
|
|
audio_out_codec = NULL;
|
|
audio_out_ctx = NULL;
|
|
audio_out_stream = NULL;
|
|
return false;
|
|
}
|
|
|
|
Debug(1,
|
|
"Audio out bit_rate (%d) sample_rate(%d) channels(%d) fmt(%d) "
|
|
"layout(%d) frame_size(%d)",
|
|
audio_out_ctx->bit_rate, audio_out_ctx->sample_rate,
|
|
audio_out_ctx->channels, audio_out_ctx->sample_fmt,
|
|
audio_out_ctx->channel_layout, audio_out_ctx->frame_size);
|
|
|
|
/** Create a new frame to store the audio samples. */
|
|
if (!(in_frame = zm_av_frame_alloc())) {
|
|
Error("Could not allocate in frame");
|
|
return false;
|
|
}
|
|
|
|
/** Create a new frame to store the audio samples. */
|
|
if (!(out_frame = zm_av_frame_alloc())) {
|
|
Error("Could not allocate out frame");
|
|
av_frame_free(&in_frame);
|
|
return false;
|
|
}
|
|
|
|
// Setup the audio resampler
|
|
resample_ctx = avresample_alloc_context();
|
|
if (!resample_ctx) {
|
|
Error("Could not allocate resample ctx\n");
|
|
return false;
|
|
}
|
|
|
|
// Some formats (i.e. WAV) do not produce the proper channel layout
|
|
if (audio_in_ctx->channel_layout == 0) {
|
|
uint64_t layout = av_get_channel_layout("mono");
|
|
av_opt_set_int(resample_ctx, "in_channel_layout",
|
|
av_get_channel_layout("mono"), 0);
|
|
Debug(1, "Bad channel layout. Need to set it to mono (%d).", layout);
|
|
} else {
|
|
av_opt_set_int(resample_ctx, "in_channel_layout",
|
|
audio_in_ctx->channel_layout, 0);
|
|
}
|
|
|
|
av_opt_set_int(resample_ctx, "in_sample_fmt",
|
|
audio_in_ctx->sample_fmt, 0);
|
|
av_opt_set_int(resample_ctx, "in_sample_rate",
|
|
audio_in_ctx->sample_rate, 0);
|
|
av_opt_set_int(resample_ctx, "in_channels", audio_in_ctx->channels,
|
|
0);
|
|
// av_opt_set_int( resample_ctx, "out_channel_layout",
|
|
// audio_out_ctx->channel_layout, 0);
|
|
av_opt_set_int(resample_ctx, "out_channel_layout",
|
|
av_get_channel_layout("mono"), 0);
|
|
av_opt_set_int(resample_ctx, "out_sample_fmt",
|
|
audio_out_ctx->sample_fmt, 0);
|
|
av_opt_set_int(resample_ctx, "out_sample_rate",
|
|
audio_out_ctx->sample_rate, 0);
|
|
av_opt_set_int(resample_ctx, "out_channels",
|
|
audio_out_ctx->channels, 0);
|
|
|
|
ret = avresample_open(resample_ctx);
|
|
if (ret < 0) {
|
|
Error("Could not open resample ctx\n");
|
|
return false;
|
|
}
|
|
|
|
#if 0
|
|
/**
|
|
* Allocate as many pointers as there are audio channels.
|
|
* Each pointer will later point to the audio samples of the corresponding
|
|
* channels (although it may be NULL for interleaved formats).
|
|
*/
|
|
if (!( converted_in_samples = reinterpret_cast<uint8_t *>calloc( audio_out_ctx->channels, sizeof(*converted_in_samples))) ) {
|
|
Error("Could not allocate converted in sample pointers\n");
|
|
return;
|
|
}
|
|
/**
|
|
* Allocate memory for the samples of all channels in one consecutive
|
|
* block for convenience.
|
|
*/
|
|
if ( (ret = av_samples_alloc( &converted_in_samples, NULL,
|
|
audio_out_ctx->channels,
|
|
audio_out_ctx->frame_size,
|
|
audio_out_ctx->sample_fmt, 0)) < 0 ) {
|
|
Error("Could not allocate converted in samples (error '%s')\n",
|
|
av_make_error_string(ret).c_str());
|
|
|
|
av_freep(converted_in_samples);
|
|
free(converted_in_samples);
|
|
return;
|
|
}
|
|
#endif
|
|
|
|
out_frame->nb_samples = audio_out_ctx->frame_size;
|
|
out_frame->format = audio_out_ctx->sample_fmt;
|
|
out_frame->channel_layout = audio_out_ctx->channel_layout;
|
|
|
|
// The codec gives us the frame size, in samples, we calculate the size of the
|
|
// samples buffer in bytes
|
|
unsigned int audioSampleBuffer_size = av_samples_get_buffer_size(
|
|
NULL, audio_out_ctx->channels, audio_out_ctx->frame_size,
|
|
audio_out_ctx->sample_fmt, 0);
|
|
converted_in_samples = (uint8_t *)av_malloc(audioSampleBuffer_size);
|
|
|
|
if (!converted_in_samples) {
|
|
Error("Could not allocate converted in sample pointers\n");
|
|
return false;
|
|
}
|
|
|
|
// Setup the data pointers in the AVFrame
|
|
if (avcodec_fill_audio_frame(out_frame, audio_out_ctx->channels,
|
|
audio_out_ctx->sample_fmt,
|
|
(const uint8_t *)converted_in_samples,
|
|
audioSampleBuffer_size, 0) < 0) {
|
|
Error("Could not allocate converted in sample pointers\n");
|
|
return false;
|
|
}
|
|
|
|
return true;
|
|
#else
|
|
Error(
|
|
"Not built with libavresample library. Cannot do audio conversion to "
|
|
"AAC");
|
|
return false;
|
|
#endif
|
|
} // end bool VideoStore::setup_resampler()
|
|
|
|
void VideoStore::dumpPacket(AVPacket *pkt) {
|
|
char b[10240];
|
|
|
|
snprintf(b, sizeof(b),
|
|
" pts: %" PRId64 ", dts: %" PRId64
|
|
", data: %p, size: %d, sindex: %d, dflags: %04x, s-pos: %" PRId64
|
|
", c-duration: %d\n",
|
|
pkt->pts, pkt->dts, pkt->data, pkt->size, pkt->stream_index,
|
|
pkt->flags, pkt->pos, pkt->duration);
|
|
Debug(1, "%s:%d:DEBUG: %s", __FILE__, __LINE__, b);
|
|
}
|
|
|
|
int VideoStore::writeVideoFramePacket(AVPacket *ipkt) {
|
|
av_init_packet(&opkt);
|
|
|
|
opkt.pts = video_next_pts;
|
|
opkt.dts = video_next_dts;
|
|
opkt.duration = 0;
|
|
|
|
int duration;
|
|
if (!video_last_pts) {
|
|
duration = 0;
|
|
} else {
|
|
duration =
|
|
av_rescale_q(ipkt->pts - video_last_pts, video_in_stream->time_base,
|
|
video_out_stream->time_base);
|
|
Debug(1, "duration calc: pts(%d) - last_pts(%d) = (%d)", ipkt->pts,
|
|
video_last_pts, duration);
|
|
if (duration < 0) {
|
|
duration = ipkt->duration;
|
|
}
|
|
}
|
|
|
|
//#if ( 0 && video_last_pts && ( ipkt->duration == AV_NOPTS_VALUE || !
|
|
//ipkt->duration ) ) {
|
|
// Video packets don't really have a duration. Audio does.
|
|
// opkt.duration = av_rescale_q(duration, video_in_stream->time_base,
|
|
// video_out_stream->time_base);
|
|
// opkt.duration = 0;
|
|
//} else {
|
|
// duration = opkt.duration = av_rescale_q(ipkt->duration,
|
|
// video_in_stream->time_base, video_out_stream->time_base);
|
|
//}
|
|
video_last_pts = ipkt->pts;
|
|
video_last_dts = ipkt->dts;
|
|
|
|
#if 0
|
|
//Scale the PTS of the outgoing packet to be the correct time base
|
|
if ( ipkt->pts != AV_NOPTS_VALUE ) {
|
|
|
|
if ( ! video_last_pts ) {
|
|
// This is the first packet.
|
|
opkt.pts = 0;
|
|
Debug(2, "Starting video video_last_pts will become (%d)", ipkt->pts);
|
|
} else {
|
|
if ( ipkt->pts < video_last_pts ) {
|
|
Debug(1, "Resetting video_last_pts from (%d) to (%d)", video_last_pts, ipkt->pts);
|
|
// wrap around, need to figure out the distance FIXME having this wrong should cause a jump, but then play ok?
|
|
opkt.pts = video_next_pts + av_rescale_q( ipkt->pts, video_in_stream->time_base, video_out_stream->time_base);
|
|
} else {
|
|
opkt.pts = video_next_pts + av_rescale_q( ipkt->pts - video_last_pts, video_in_stream->time_base, video_out_stream->time_base);
|
|
}
|
|
}
|
|
Debug(3, "opkt.pts = %d from ipkt->pts(%d) - last_pts(%d)", opkt.pts, ipkt->pts, video_last_pts);
|
|
video_last_pts = ipkt->pts;
|
|
} else {
|
|
Debug(3, "opkt.pts = undef");
|
|
opkt.pts = AV_NOPTS_VALUE;
|
|
}
|
|
// Just because the in stream wraps, doesn't mean the out needs to. Really, if we are limiting ourselves to 10min segments I can't imagine every wrapping in the out. So need to handle in wrap, without causing out wrap.
|
|
if ( !video_last_dts ) {
|
|
// This is the first packet.
|
|
opkt.dts = 0;
|
|
Debug(1, "Starting video video_last_dts will become (%lu)", ipkt->dts);
|
|
video_last_dts = ipkt->dts;
|
|
} else {
|
|
// Scale the DTS of the outgoing packet to be the correct time base
|
|
|
|
if ( ipkt->dts == AV_NOPTS_VALUE ) {
|
|
// why are we using cur_dts instead of packet.dts? I think cur_dts is in AV_TIME_BASE_Q, but ipkt.dts is in video_in_stream->time_base
|
|
if ( video_in_stream->cur_dts < video_last_dts ) {
|
|
Debug(1, "Resetting video_last_dts from (%d) to (%d) p.dts was (%d)", video_last_dts, video_in_stream->cur_dts, ipkt->dts);
|
|
opkt.dts = video_next_dts + av_rescale_q(video_in_stream->cur_dts, AV_TIME_BASE_Q, video_out_stream->time_base);
|
|
} else {
|
|
opkt.dts = video_next_dts + av_rescale_q(video_in_stream->cur_dts - video_last_dts, AV_TIME_BASE_Q, video_out_stream->time_base);
|
|
}
|
|
Debug(3, "opkt.dts = %d from video_in_stream->cur_dts(%d) - previus_dts(%d)", opkt.dts, video_in_stream->cur_dts, video_last_dts);
|
|
video_last_dts = video_in_stream->cur_dts;
|
|
} else {
|
|
if ( ipkt->dts < video_last_dts ) {
|
|
Debug(1, "Resetting video_last_dts from (%d) to (%d)", video_last_dts, ipkt->dts);
|
|
opkt.dts = video_next_dts + av_rescale_q( ipkt->dts, video_in_stream->time_base, video_out_stream->time_base);
|
|
} else {
|
|
opkt.dts = video_next_dts + av_rescale_q( ipkt->dts - video_last_dts, video_in_stream->time_base, video_out_stream->time_base);
|
|
}
|
|
Debug(3, "opkt.dts = %d from ipkt.dts(%d) - previus_dts(%d)", opkt.dts, ipkt->dts, video_last_dts);
|
|
video_last_dts = ipkt->dts;
|
|
}
|
|
}
|
|
#endif
|
|
if (opkt.dts > opkt.pts) {
|
|
Debug(1,
|
|
"opkt.dts(%d) must be <= opkt.pts(%d). Decompression must happen "
|
|
"before presentation.",
|
|
opkt.dts, opkt.pts);
|
|
opkt.dts = opkt.pts;
|
|
}
|
|
|
|
opkt.flags = ipkt->flags;
|
|
int keyframe = opkt.flags & AV_PKT_FLAG_KEY;
|
|
opkt.pos = -1;
|
|
|
|
opkt.data = ipkt->data;
|
|
opkt.size = ipkt->size;
|
|
|
|
opkt.stream_index = video_out_stream->index;
|
|
|
|
AVPacket safepkt;
|
|
memcpy(&safepkt, &opkt, sizeof(AVPacket));
|
|
|
|
Debug(1,
|
|
"writing video packet keyframe(%d) pts(%d) dts(%d) duration(%d) "
|
|
"ipkt.duration(%d)",
|
|
keyframe, opkt.pts, opkt.dts, duration, ipkt->duration);
|
|
if ((opkt.data == NULL) || (opkt.size < 1)) {
|
|
Warning("%s:%d: Mangled AVPacket: discarding frame", __FILE__, __LINE__);
|
|
dumpPacket(ipkt);
|
|
dumpPacket(&opkt);
|
|
|
|
} else if ((video_next_dts > 0) && (video_next_dts > opkt.dts)) {
|
|
Warning("%s:%d: DTS out of order: %lld \u226E %lld; discarding frame",
|
|
__FILE__, __LINE__, video_next_dts, opkt.dts);
|
|
video_next_dts = opkt.dts;
|
|
dumpPacket(&opkt);
|
|
|
|
} else {
|
|
video_next_dts = opkt.dts + duration;
|
|
video_next_pts = opkt.pts + duration;
|
|
ret = av_interleaved_write_frame(oc, &opkt);
|
|
if (ret < 0) {
|
|
// There's nothing we can really do if the frame is rejected, just drop it
|
|
// and get on with the next
|
|
Warning(
|
|
"%s:%d: Writing frame [av_interleaved_write_frame()] failed: %s(%d) "
|
|
" ",
|
|
__FILE__, __LINE__, av_make_error_string(ret).c_str(), (ret));
|
|
dumpPacket(&safepkt);
|
|
#if LIBAVCODEC_VERSION_CHECK(57, 64, 0, 64, 0)
|
|
zm_dump_codecpar(video_in_stream->codecpar);
|
|
zm_dump_codecpar(video_out_stream->codecpar);
|
|
#endif
|
|
}
|
|
}
|
|
|
|
zm_av_packet_unref(&opkt);
|
|
|
|
return 0;
|
|
} // end int VideoStore::writeVideoFramePacket( AVPacket *ipkt )
|
|
|
|
int VideoStore::writeAudioFramePacket(AVPacket *ipkt) {
|
|
Debug(4, "writeAudioFrame");
|
|
|
|
if (!audio_out_stream) {
|
|
Debug(1, "Called writeAudioFramePacket when no audio_out_stream");
|
|
return 0; // FIXME -ve return codes do not free packet in ffmpeg_camera at
|
|
// the moment
|
|
}
|
|
|
|
if (audio_out_codec) {
|
|
Debug(3, "Have audio codec");
|
|
#ifdef HAVE_LIBAVRESAMPLE
|
|
|
|
#if LIBAVCODEC_VERSION_CHECK(57, 64, 0, 64, 0)
|
|
ret = avcodec_send_packet(audio_in_ctx, ipkt);
|
|
if (ret < 0) {
|
|
Error("avcodec_send_packet fail %s", av_make_error_string(ret).c_str());
|
|
return 0;
|
|
}
|
|
|
|
ret = avcodec_receive_frame(audio_in_ctx, in_frame);
|
|
if (ret < 0) {
|
|
Error("avcodec_receive_frame fail %s", av_make_error_string(ret).c_str());
|
|
return 0;
|
|
}
|
|
Debug(2,
|
|
"Input Frame: samples(%d), format(%d), sample_rate(%d), channel "
|
|
"layout(%d)",
|
|
in_frame->nb_samples, in_frame->format,
|
|
in_frame->sample_rate, in_frame->channel_layout);
|
|
#else
|
|
/**
|
|
* Decode the audio frame stored in the packet.
|
|
* The in audio stream decoder is used to do this.
|
|
* If we are at the end of the file, pass an empty packet to the decoder
|
|
* to flush it.
|
|
*/
|
|
if ((ret = avcodec_decode_audio4(audio_in_ctx, in_frame,
|
|
&data_present, ipkt)) < 0) {
|
|
Error("Could not decode frame (error '%s')\n",
|
|
av_make_error_string(ret).c_str());
|
|
dumpPacket(ipkt);
|
|
av_frame_free(&in_frame);
|
|
return 0;
|
|
}
|
|
if (!data_present) {
|
|
Debug(2, "Not ready to transcode a frame yet.");
|
|
return 0;
|
|
}
|
|
#endif
|
|
int frame_size = out_frame->nb_samples;
|
|
|
|
// Resample the in into the audioSampleBuffer until we proceed the whole
|
|
// decoded data
|
|
if ((ret =
|
|
avresample_convert(resample_ctx, NULL, 0, 0, in_frame->data,
|
|
0, in_frame->nb_samples)) < 0) {
|
|
Error("Could not resample frame (error '%s')\n",
|
|
av_make_error_string(ret).c_str());
|
|
av_frame_unref(in_frame);
|
|
return 0;
|
|
}
|
|
av_frame_unref(in_frame);
|
|
|
|
int samples_available = avresample_available(resample_ctx);
|
|
|
|
if (samples_available < frame_size) {
|
|
Debug(1, "Not enough samples yet (%d)", samples_available);
|
|
return 0;
|
|
}
|
|
|
|
Debug(3, "Output_frame samples (%d)", out_frame->nb_samples);
|
|
// Read a frame audio data from the resample fifo
|
|
if (avresample_read(resample_ctx, out_frame->data, frame_size) !=
|
|
frame_size) {
|
|
Warning("Error reading resampled audio: ");
|
|
return 0;
|
|
}
|
|
Debug(2,
|
|
"Frame: samples(%d), format(%d), sample_rate(%d), channel layout(%d)",
|
|
out_frame->nb_samples, out_frame->format,
|
|
out_frame->sample_rate, out_frame->channel_layout);
|
|
|
|
av_init_packet(&opkt);
|
|
Debug(5, "after init packet");
|
|
|
|
#if LIBAVCODEC_VERSION_CHECK(57, 64, 0, 64, 0)
|
|
if ((ret = avcodec_send_frame(audio_out_ctx, out_frame)) < 0) {
|
|
Error("Could not send frame (error '%s')",
|
|
av_make_error_string(ret).c_str());
|
|
zm_av_packet_unref(&opkt);
|
|
return 0;
|
|
}
|
|
|
|
// av_frame_unref( out_frame );
|
|
|
|
if ((ret = avcodec_receive_packet(audio_out_ctx, &opkt)) < 0) {
|
|
if (AVERROR(EAGAIN) == ret) {
|
|
// THe codec may need more samples than it has, perfectly valid
|
|
Debug(3, "Could not recieve packet (error '%s')",
|
|
av_make_error_string(ret).c_str());
|
|
} else {
|
|
Error("Could not recieve packet (error %d = '%s')", ret,
|
|
av_make_error_string(ret).c_str());
|
|
}
|
|
zm_av_packet_unref(&opkt);
|
|
av_frame_unref(in_frame);
|
|
// av_frame_unref( out_frame );
|
|
return 0;
|
|
}
|
|
#else
|
|
if ((ret = avcodec_encode_audio2(audio_out_ctx, &opkt, out_frame,
|
|
&data_present)) < 0) {
|
|
Error("Could not encode frame (error '%s')",
|
|
av_make_error_string(ret).c_str());
|
|
zm_av_packet_unref(&opkt);
|
|
return 0;
|
|
}
|
|
if (!data_present) {
|
|
Debug(2, "Not ready to out a frame yet.");
|
|
zm_av_packet_unref(&opkt);
|
|
return 0;
|
|
}
|
|
#endif
|
|
|
|
#endif
|
|
} else {
|
|
av_init_packet(&opkt);
|
|
Debug(5, "after init packet");
|
|
opkt.data = ipkt->data;
|
|
opkt.size = ipkt->size;
|
|
}
|
|
|
|
// PTS is difficult, because of the buffering of the audio packets in the
|
|
// resampler. So we have to do it once we actually have a packet...
|
|
// audio_last_pts is the pts of ipkt, audio_next_pts is the last pts of the
|
|
// out
|
|
|
|
// Scale the PTS of the outgoing packet to be the correct time base
|
|
#if 0
|
|
if ( ipkt->pts != AV_NOPTS_VALUE ) {
|
|
if ( !audio_last_pts ) {
|
|
opkt.pts = 0;
|
|
Debug(1, "No audio_last_pts");
|
|
} else {
|
|
if ( audio_last_pts > ipkt->pts ) {
|
|
Debug(1, "Resetting audio_start_pts from (%d) to (%d)", audio_last_pts, ipkt->pts);
|
|
opkt.pts = audio_next_pts + av_rescale_q(ipkt->pts, audio_in_stream->time_base, audio_out_stream->time_base);
|
|
} else {
|
|
opkt.pts = audio_next_pts + av_rescale_q(ipkt->pts - audio_last_pts, audio_in_stream->time_base, audio_out_stream->time_base);
|
|
}
|
|
Debug(2, "audio opkt.pts = %d from ipkt->pts(%d) - last_pts(%d)", opkt.pts, ipkt->pts, audio_last_pts);
|
|
}
|
|
audio_last_pts = ipkt->pts;
|
|
} else {
|
|
Debug(2, "opkt.pts = undef");
|
|
opkt.pts = AV_NOPTS_VALUE;
|
|
}
|
|
#else
|
|
opkt.pts = audio_next_pts;
|
|
opkt.dts = audio_next_dts;
|
|
#endif
|
|
|
|
#if 0
|
|
if ( ipkt->dts == AV_NOPTS_VALUE ) {
|
|
// So if the in has no dts assigned... still need an out dts... so we use cur_dts?
|
|
|
|
if ( audio_last_dts >= audio_in_stream->cur_dts ) {
|
|
Debug(1, "Resetting audio_last_dts from (%d) to cur_dts (%d)", audio_last_dts, audio_in_stream->cur_dts);
|
|
opkt.dts = audio_next_dts + av_rescale_q( audio_in_stream->cur_dts, AV_TIME_BASE_Q, audio_out_stream->time_base);
|
|
} else {
|
|
opkt.dts = audio_next_dts + av_rescale_q( audio_in_stream->cur_dts - audio_last_dts, AV_TIME_BASE_Q, audio_out_stream->time_base);
|
|
}
|
|
audio_last_dts = audio_in_stream->cur_dts;
|
|
Debug(2, "opkt.dts = %d from video_in_stream->cur_dts(%d) - last_dts(%d)", opkt.dts, audio_in_stream->cur_dts, audio_last_dts);
|
|
} else {
|
|
if ( audio_last_dts >= ipkt->dts ) {
|
|
Debug(1, "Resetting audio_last_dts from (%d) to (%d)", audio_last_dts, ipkt->dts );
|
|
opkt.dts = audio_next_dts + av_rescale_q(ipkt->dts, audio_in_stream->time_base, audio_out_stream->time_base);
|
|
} else {
|
|
opkt.dts = audio_next_dts + av_rescale_q(ipkt->dts - audio_last_dts, audio_in_stream->time_base, audio_out_stream->time_base);
|
|
Debug(2, "opkt.dts = %d from previous(%d) + ( ipkt->dts(%d) - last_dts(%d) )", opkt.dts, audio_next_dts, ipkt->dts, audio_last_dts );
|
|
}
|
|
}
|
|
}
|
|
#endif
|
|
// audio_last_dts = ipkt->dts;
|
|
if (opkt.dts > opkt.pts) {
|
|
Debug(1,
|
|
"opkt.dts(%d) must be <= opkt.pts(%d). Decompression must happen "
|
|
"before presentation.",
|
|
opkt.dts, opkt.pts);
|
|
opkt.dts = opkt.pts;
|
|
}
|
|
|
|
// I wonder if we could just use duration instead of all the hoop jumping
|
|
// above?
|
|
//
|
|
if (out_frame) {
|
|
opkt.duration = out_frame->nb_samples;
|
|
} else {
|
|
opkt.duration = ipkt->duration;
|
|
}
|
|
// opkt.duration = av_rescale_q(ipkt->duration, audio_in_stream->time_base,
|
|
// audio_out_stream->time_base);
|
|
Debug(2, "opkt.pts (%d), opkt.dts(%d) opkt.duration = (%d)", opkt.pts,
|
|
opkt.dts, opkt.duration);
|
|
|
|
// pkt.pos: byte position in stream, -1 if unknown
|
|
opkt.pos = -1;
|
|
opkt.stream_index = audio_out_stream->index;
|
|
audio_next_dts = opkt.dts + opkt.duration;
|
|
audio_next_pts = opkt.pts + opkt.duration;
|
|
|
|
AVPacket safepkt;
|
|
memcpy(&safepkt, &opkt, sizeof(AVPacket));
|
|
ret = av_interleaved_write_frame(oc, &opkt);
|
|
if (ret != 0) {
|
|
Error("Error writing audio frame packet: %s\n",
|
|
av_make_error_string(ret).c_str());
|
|
dumpPacket(&safepkt);
|
|
} else {
|
|
Debug(2, "Success writing audio frame");
|
|
}
|
|
zm_av_packet_unref(&opkt);
|
|
return 0;
|
|
} // end int VideoStore::writeAudioFramePacket( AVPacket *ipkt )
|