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zoneminder/src/zm_videostore.cpp

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// ZoneMinder Video Storage Implementation
// Written by Chris Wiggins
// http://chriswiggins.co.nz
// Modification by Steve Gilvarry
//
// 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.
//
#define __STDC_FORMAT_MACROS 1
#include <stdlib.h>
#include <string.h>
#include <inttypes.h>
#include "zm.h"
#include "zm_videostore.h"
extern "C" {
#include "libavutil/time.h"
}
VideoStore::VideoStore(const char *filename_in, const char *format_in,
AVStream *p_video_input_stream,
AVStream *p_audio_input_stream,
int64_t nStartTime,
Monitor * monitor
) {
video_input_stream = p_video_input_stream;
audio_input_stream = p_audio_input_stream;
#if LIBAVCODEC_VERSION_CHECK(57, 0, 0, 0, 0)
video_input_context = avcodec_alloc_context3( NULL );
avcodec_parameters_to_context( video_input_context, video_input_stream->codecpar );
#else
video_input_context = video_input_stream->codec;
#endif
//store inputs in variables local to class
filename = filename_in;
format = format_in;
Info("Opening video storage stream %s format: %s\n", filename, format);
ret = avformat_alloc_output_context2(&oc, NULL, NULL, filename);
if ( ret < 0 ) {
Warning("Could not create video storage stream %s as no output context"
" could be assigned based on filename: %s",
filename,
av_make_error_string(ret).c_str()
);
} else {
Debug(2, "Success allocating output context");
}
//Couldn't deduce format from filename, trying from format name
if ( ! oc ) {
avformat_alloc_output_context2(&oc, NULL, format, filename);
if (!oc) {
Fatal("Could not create video storage stream %s as no output context"
" could not be assigned based on filename or format %s",
filename, format);
}
} else {
Debug(2, "Success alocateing output context");
}
AVDictionary *pmetadata = NULL;
int dsr = av_dict_set(&pmetadata, "title", "Zoneminder Security Recording", 0);
if (dsr < 0) Warning("%s:%d: title set failed", __FILE__, __LINE__ );
oc->metadata = pmetadata;
output_format = oc->oformat;
#if LIBAVCODEC_VERSION_CHECK(57, 0, 0, 0, 0)
// Since we are not re-encoding, all we have to do is copy the parameters
video_output_context = avcodec_alloc_context3( NULL );
// Copy params from inputstream to context
ret = avcodec_parameters_to_context( video_output_context, video_input_stream->codecpar );
if ( ret < 0 ) {
Error( "Could not initialize context parameteres");
return;
} else {
Debug( 2, "Success getting parameters");
}
video_output_stream = avformat_new_stream( oc, NULL );
if ( ! video_output_stream ) {
Fatal("Unable to create video out stream\n");
} else {
Debug(2, "Success creating video out stream" );
}
// Now copy them to the output stream
ret = avcodec_parameters_from_context( video_output_stream->codecpar, video_output_context );
if ( ret < 0 ) {
Error( "Could not initialize stream parameteres");
return;
} else {
Debug(2, "Success setting parameters");
}
zm_dump_stream_format( oc, 0, 0, 1 );
#else
video_output_stream = avformat_new_stream(oc, (AVCodec*)video_input_context->codec );
if ( ! video_output_stream ) {
Fatal("Unable to create video out stream\n");
} else {
Debug(2, "Success creating video out stream" );
}
video_output_context = video_output_stream->codec;
ret = avcodec_copy_context( video_output_context, video_input_context );
if (ret < 0) {
Fatal("Unable to copy input video context to output video context %s\n",
av_make_error_string(ret).c_str());
} else {
Debug(3, "Success copying context" );
}
#endif
// Just copy them from the input, no reason to choose different
video_output_context->time_base = video_input_context->time_base;
video_output_stream->time_base = video_input_stream->time_base;
Debug(3, "Time bases: VIDEO input stream (%d/%d) input codec: (%d/%d) output stream: (%d/%d) output codec (%d/%d)",
video_input_stream->time_base.num,
video_input_stream->time_base.den,
video_input_context->time_base.num,
video_input_context->time_base.den,
video_output_stream->time_base.num,
video_output_stream->time_base.den,
video_output_context->time_base.num,
video_output_context->time_base.den
);
// WHY?
//video_output_context->codec_tag = 0;
if ( ! video_output_context->codec_tag ) {
Debug(2, "No codec_tag");
if (! oc->oformat->codec_tag
|| av_codec_get_id (oc->oformat->codec_tag, video_input_context->codec_tag) == video_output_context->codec_id
|| av_codec_get_tag(oc->oformat->codec_tag, video_input_context->codec_id) <= 0) {
Warning("Setting codec tag");
video_output_context->codec_tag = video_input_context->codec_tag;
}
}
if (oc->oformat->flags & AVFMT_GLOBALHEADER) {
video_output_context->flags |= CODEC_FLAG_GLOBAL_HEADER;
}
Monitor::Orientation orientation = monitor->getOrientation();
Debug(3, "Have orientation" );
if ( orientation ) {
if ( orientation == Monitor::ROTATE_0 ) {
} else if ( orientation == Monitor::ROTATE_90 ) {
dsr = av_dict_set( &video_output_stream->metadata, "rotate", "90", 0);
if (dsr < 0) Warning("%s:%d: title set failed", __FILE__, __LINE__ );
} else if ( orientation == Monitor::ROTATE_180 ) {
dsr = av_dict_set( &video_output_stream->metadata, "rotate", "180", 0);
if (dsr < 0) Warning("%s:%d: title set failed", __FILE__, __LINE__ );
} else if ( orientation == Monitor::ROTATE_270 ) {
dsr = av_dict_set( &video_output_stream->metadata, "rotate", "270", 0);
if (dsr < 0) Warning("%s:%d: title set failed", __FILE__, __LINE__ );
} else {
Warning( "Unsupported Orientation(%d)", orientation );
}
}
audio_output_codec = NULL;
audio_input_context = NULL;
audio_output_stream = NULL;
#ifdef HAVE_LIBAVRESAMPLE
resample_context = NULL;
#endif
if ( audio_input_stream ) {
Debug(3, "Have audio stream" );
#if LIBAVCODEC_VERSION_CHECK(57, 0, 0, 0, 0)
audio_input_context = avcodec_alloc_context3( NULL );
ret = avcodec_parameters_to_context( audio_input_context, audio_input_stream->codecpar );
#else
audio_input_context = audio_input_stream->codec;
#endif
if ( audio_input_context->codec_id != AV_CODEC_ID_AAC ) {
static char error_buffer[256];
avcodec_string(error_buffer, sizeof(error_buffer), audio_input_context, 0 );
Debug(2, "Got something other than AAC (%s)", error_buffer );
if ( ! setup_resampler() ) {
return;
}
} else {
Debug(3, "Got AAC" );
audio_output_stream = avformat_new_stream(oc, (AVCodec*)audio_input_context->codec);
if ( ! audio_output_stream ) {
Error("Unable to create audio out stream\n");
audio_output_stream = NULL;
} else {
Debug(2, "setting parameters");
#if LIBAVCODEC_VERSION_CHECK(57, 0, 0, 0, 0)
audio_output_context = avcodec_alloc_context3( NULL );
// Copy params from inputstream to context
ret = avcodec_parameters_to_context( audio_output_context, audio_input_stream->codecpar );
#else
audio_output_context = audio_output_stream->codec;
ret = avcodec_copy_context(audio_output_context, audio_input_context);
#endif
if (ret < 0) {
Error("Unable to copy audio context %s\n", av_make_error_string(ret).c_str());
audio_output_stream = NULL;
} else {
audio_output_context->codec_tag = 0;
if ( audio_output_context->channels > 1 ) {
Warning("Audio isn't mono, changing it.");
audio_output_context->channels = 1;
} else {
Debug(3, "Audio is mono");
}
}
} // end if audio_output_stream
} // end if is AAC
if ( audio_output_stream ) {
if (oc->oformat->flags & AVFMT_GLOBALHEADER) {
audio_output_context->flags |= CODEC_FLAG_GLOBAL_HEADER;
}
}
} // end if audio_input_stream
/* open the output file, if needed */
if (!(output_format->flags & AVFMT_NOFILE)) {
ret = avio_open2(&oc->pb, filename, AVIO_FLAG_WRITE,NULL,NULL);
if (ret < 0) {
Fatal("Could not open output file '%s': %s\n", filename,
av_make_error_string(ret).c_str());
}
}
//os->ctx_inited = 1;
//avio_flush(ctx->pb);
//av_dict_free(&opts);
zm_dump_stream_format( oc, 0, 0, 1 );
if ( audio_output_stream )
zm_dump_stream_format( oc, 1, 0, 1 );
AVDictionary * opts = NULL;
//av_dict_set(&opts, "movflags", "frag_custom+dash+delay_moov", 0);
//av_dict_set(&opts, "movflags", "frag_custom+dash+delay_moov", 0);
//av_dict_set(&opts, "movflags", "frag_keyframe+empty_moov+default_base_moof", 0);
if ((ret = avformat_write_header(oc, NULL)) < 0) {
//if ((ret = avformat_write_header(oc, &opts)) < 0) {
Warning("Unable to set movflags to frag_custom+dash+delay_moov");
/* Write the stream header, if any. */
ret = avformat_write_header(oc, NULL);
} else if (av_dict_count(opts) != 0) {
Warning("some options not set\n");
}
if (ret < 0) {
Error("Error occurred when writing output file header to %s: %s\n",
filename,
av_make_error_string(ret).c_str());
}
if ( opts )
av_dict_free(&opts);
video_last_pts = 0;
video_last_dts = 0;
audio_last_pts = 0;
audio_last_dts = 0;
video_previous_pts = 0;
video_previous_dts = 0;
audio_previous_pts = 0;
audio_previous_dts = 0;
} // VideoStore::VideoStore
VideoStore::~VideoStore(){
if ( audio_output_codec ) {
// Do we need to flush the outputs? I have no idea.
AVPacket pkt;
int got_packet = 0;
av_init_packet(&pkt);
pkt.data = NULL;
pkt.size = 0;
int64_t size;
while(1) {
#if LIBAVCODEC_VERSION_CHECK(57, 0, 0, 0, 0)
ret = avcodec_receive_packet( audio_output_context, &pkt );
#else
ret = avcodec_encode_audio2( audio_output_context, &pkt, NULL, &got_packet );
#endif
if (ret < 0) {
Error("ERror encoding audio while flushing");
break;
}
Debug(1, "Have audio encoder, need to flush it's output" );
size += pkt.size;
if (!got_packet) {
break;
}
Debug(2, "writing flushed packet pts(%d) dts(%d) duration(%d)", pkt.pts, pkt.dts, pkt.duration );
if (pkt.pts != AV_NOPTS_VALUE)
pkt.pts = av_rescale_q(pkt.pts, audio_output_context->time_base, audio_output_stream->time_base);
if (pkt.dts != AV_NOPTS_VALUE)
pkt.dts = av_rescale_q(pkt.dts, audio_output_context->time_base, audio_output_stream->time_base);
if (pkt.duration > 0)
pkt.duration = av_rescale_q(pkt.duration, audio_output_context->time_base, audio_output_stream->time_base);
Debug(2, "writing flushed packet pts(%d) dts(%d) duration(%d)", pkt.pts, pkt.dts, pkt.duration );
pkt.stream_index = audio_output_stream->index;
av_interleaved_write_frame( oc, &pkt );
zm_av_packet_unref( &pkt );
} // while 1
}
// Flush Queues
av_interleaved_write_frame( oc, NULL );
/* Write the trailer before close */
if ( int rc = av_write_trailer(oc) ) {
Error("Error writing trailer %s", av_err2str( rc ) );
} else {
Debug(3, "Sucess Writing trailer");
}
// I wonder if we should be closing the file first.
// I also wonder if we really need to be doing all the context allocation/de-allocation constantly, or whether we can just re-use it. Just do a file open/close/writeheader/etc.
// What if we were only doing audio recording?
if ( video_output_stream ) {
avcodec_close(video_output_context);
}
if (audio_output_stream) {
avcodec_close(audio_output_context);
#ifdef HAVE_LIBAVRESAMPLE
if ( resample_context ) {
avresample_close( resample_context );
avresample_free( &resample_context );
}
#endif
}
// WHen will be not using a file ?
if (!(output_format->flags & AVFMT_NOFILE)) {
/* Close the output file. */
if ( int rc = avio_close(oc->pb) ) {
Error("Error closing avio %s", av_err2str( rc ) );
}
} else {
Debug(3, "Not closing avio because we are not writing to a file.");
}
/* free the stream */
avformat_free_context(oc);
}
bool VideoStore::setup_resampler() {
#ifdef HAVE_LIBAVRESAMPLE
static char error_buffer[256];
#if LIBAVCODEC_VERSION_CHECK(57, 0, 0, 0, 0)
// Newer ffmpeg wants to keep everything separate... so have to lookup our own decoder, can't reuse the one from the camera.
AVCodec *audio_input_codec = avcodec_find_decoder(audio_input_stream->codecpar->codec_id);
#else
AVCodec *audio_input_codec = avcodec_find_decoder(audio_input_context->codec_id);
#endif
ret = avcodec_open2( audio_input_context, audio_input_codec, NULL );
if ( ret < 0 ) {
Error("Can't open input codec!");
return false;
}
audio_output_codec = avcodec_find_encoder(AV_CODEC_ID_AAC);
if ( ! audio_output_codec ) {
Error("Could not find codec for AAC");
return false;
}
Debug(2, "Have audio output codec");
//audio_output_context = audio_output_stream->codec;
audio_output_context = avcodec_alloc_context3( audio_output_codec );
if ( ! audio_output_context ) {
Error( "could not allocate codec context for AAC\n");
audio_output_stream = NULL;
return false;
}
Debug(2, "Have audio_output_context");
/* put sample parameters */
audio_output_context->bit_rate = audio_input_context->bit_rate;
audio_output_context->sample_rate = audio_input_context->sample_rate;
audio_output_context->channels = audio_input_context->channels;
audio_output_context->channel_layout = audio_input_context->channel_layout;
audio_output_context->sample_fmt = audio_input_context->sample_fmt;
audio_output_context->refcounted_frames = 1;
if ( audio_output_codec->supported_samplerates ) {
int found = 0;
for ( unsigned int i = 0; audio_output_codec->supported_samplerates[i]; i++) {
if ( audio_output_context->sample_rate == audio_output_codec->supported_samplerates[i] ) {
found = 1;
break;
}
}
if ( found ) {
Debug(3, "Sample rate is good");
} else {
audio_output_context->sample_rate = audio_output_codec->supported_samplerates[0];
Debug(1, "Sampel rate is no good, setting to (%d)", audio_output_codec->supported_samplerates[0] );
}
}
/* check that the encoder supports s16 pcm input */
if ( ! check_sample_fmt( audio_output_codec, audio_output_context->sample_fmt ) ) {
Debug( 3, "Encoder does not support sample format %s, setting to FLTP",
av_get_sample_fmt_name( audio_output_context->sample_fmt));
audio_output_context->sample_fmt = AV_SAMPLE_FMT_FLTP;
}
audio_output_context->time_base = (AVRational){ 1, audio_output_context->sample_rate };
Debug(1, "Audio output bit_rate (%d) sample_rate(%d) channels(%d) fmt(%d) layout(%d) frame_size(%d)",
audio_output_context->bit_rate,
audio_output_context->sample_rate,
audio_output_context->channels,
audio_output_context->sample_fmt,
audio_output_context->channel_layout,
audio_output_context->frame_size
);
// Now copy them to the output stream
audio_output_stream = avformat_new_stream( oc, audio_output_codec );
#if LIBAVCODEC_VERSION_CHECK(57, 0, 0, 0, 0)
ret = avcodec_parameters_from_context( audio_output_stream->codecpar, audio_output_context );
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_output_context, audio_output_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_output_codec = NULL;
audio_output_context = NULL;
audio_output_stream = NULL;
return false;
}
/** Create a new frame to store the audio samples. */
if (!(input_frame = zm_av_frame_alloc())) {
Error("Could not allocate input frame");
return false;
}
/** Create a new frame to store the audio samples. */
if (!(output_frame = zm_av_frame_alloc())) {
Error("Could not allocate output frame");
av_frame_free( &input_frame );
return false;
}
// Setup the audio resampler
resample_context = avresample_alloc_context();
if ( ! resample_context ) {
Error( "Could not allocate resample context\n");
return false;
}
// Some formats (i.e. WAV) do not produce the proper channel layout
if ( audio_input_context->channel_layout == 0 ) {
Error( "Bad channel layout. Need to set it to mono.\n");
av_opt_set_int( resample_context, "in_channel_layout", av_get_channel_layout( "mono" ), 0 );
} else {
av_opt_set_int( resample_context, "in_channel_layout", audio_input_context->channel_layout, 0 );
}
av_opt_set_int( resample_context, "in_sample_fmt", audio_input_context->sample_fmt, 0);
av_opt_set_int( resample_context, "in_sample_rate", audio_input_context->sample_rate, 0);
av_opt_set_int( resample_context, "in_channels", audio_input_context->channels,0);
//av_opt_set_int( resample_context, "out_channel_layout", audio_output_context->channel_layout, 0);
av_opt_set_int( resample_context, "out_channel_layout", av_get_channel_layout( "mono" ), 0 );
av_opt_set_int( resample_context, "out_sample_fmt", audio_output_context->sample_fmt, 0);
av_opt_set_int( resample_context, "out_sample_rate", audio_output_context->sample_rate, 0);
av_opt_set_int( resample_context, "out_channels", audio_output_context->channels, 0);
ret = avresample_open( resample_context );
if ( ret < 0 ) {
Error( "Could not open resample context\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_input_samples = (uint8_t *)calloc( audio_output_context->channels, sizeof(*converted_input_samples))) ) {
Error( "Could not allocate converted input sample pointers\n");
return;
}
/**
* Allocate memory for the samples of all channels in one consecutive
* block for convenience.
*/
if ((ret = av_samples_alloc( &converted_input_samples, NULL,
audio_output_context->channels,
audio_output_context->frame_size,
audio_output_context->sample_fmt, 0)) < 0) {
Error( "Could not allocate converted input samples (error '%s')\n",
av_make_error_string(ret).c_str() );
av_freep(converted_input_samples);
free(converted_input_samples);
return;
}
#endif
output_frame->nb_samples = audio_output_context->frame_size;
output_frame->format = audio_output_context->sample_fmt;
output_frame->channel_layout = audio_output_context->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_output_context->channels, audio_output_context->frame_size, audio_output_context->sample_fmt, 0 );
converted_input_samples = (uint8_t*) av_malloc( audioSampleBuffer_size );
if ( !converted_input_samples ) {
Error( "Could not allocate converted input sample pointers\n");
return false;
}
// Setup the data pointers in the AVFrame
if ( avcodec_fill_audio_frame(
output_frame,
audio_output_context->channels,
audio_output_context->sample_fmt,
(const uint8_t*) converted_input_samples,
audioSampleBuffer_size, 0 ) < 0 ) {
Error( "Could not allocate converted input sample pointers\n");
return false;
}
return true;
#else
Error("Not built with libavresample library. Cannot do audio conversion to AAC");
return false;
#endif
}
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: %" PRId64 "\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);
int duration;
//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_previous_pts + av_rescale_q( ipkt->pts, video_input_stream->time_base, video_output_stream->time_base);
} else {
opkt.pts = video_previous_pts + av_rescale_q( ipkt->pts - video_last_pts, video_input_stream->time_base, video_output_stream->time_base);
}
}
Debug(3, "opkt.pts = %d from ipkt->pts(%d) - last_pts(%d)", opkt.pts, ipkt->pts, video_last_pts );
duration = ipkt->pts - video_last_pts;
video_last_pts = ipkt->pts;
} else {
Debug(3, "opkt.pts = undef");
opkt.pts = AV_NOPTS_VALUE;
}
//Scale the DTS of the outgoing packet to be the correct time base
// Just because the input stream wraps, doesn't mean the output needs to. Really, if we are limiting ourselves to 10min segments I can't imagine every wrapping in the output. So need to handle input wrap, without causing output wrap.
if ( ! video_last_dts ) {
// This is the first packet.
opkt.dts = 0;
Debug(1, "Starting video video_last_dts will become (%d)", ipkt->dts );
video_last_dts = ipkt->dts;
} else {
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_input_stream->time_base
if ( video_input_stream->cur_dts < video_last_dts ) {
Debug(1, "Resetting video_last_dts from (%d) to (%d) p.dts was (%d)", video_last_dts, video_input_stream->cur_dts, ipkt->dts );
opkt.dts = video_previous_dts + av_rescale_q(video_input_stream->cur_dts, AV_TIME_BASE_Q, video_output_stream->time_base);
} else {
opkt.dts = video_previous_dts + av_rescale_q(video_input_stream->cur_dts - video_last_dts, AV_TIME_BASE_Q, video_output_stream->time_base);
}
Debug(3, "opkt.dts = %d from video_input_stream->cur_dts(%d) - previus_dts(%d)", opkt.dts, video_input_stream->cur_dts, video_last_dts );
video_last_dts = video_input_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_previous_dts + av_rescale_q( ipkt->dts, video_input_stream->time_base, video_output_stream->time_base);
} else {
opkt.dts = video_previous_dts + av_rescale_q( ipkt->dts - video_last_dts, video_input_stream->time_base, video_output_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;
}
}
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;
}
if ( ipkt->duration == AV_NOPTS_VALUE ) {
opkt.duration = av_rescale_q( duration, video_input_stream->time_base, video_output_stream->time_base);
} else {
opkt.duration = av_rescale_q(ipkt->duration, video_input_stream->time_base, video_output_stream->time_base);
}
opkt.flags = ipkt->flags;
opkt.pos=-1;
opkt.data = ipkt->data;
opkt.size = ipkt->size;
// Some camera have audio on stream 0 and video on stream 1. So when we remove the audio, video stream has to go on 0
if ( ipkt->stream_index > 0 and ! audio_output_stream ) {
Debug(1,"Setting stream index to 0 instead of %d", ipkt->stream_index );
opkt.stream_index = 0;
} else {
opkt.stream_index = ipkt->stream_index;
}
AVPacket safepkt;
memcpy(&safepkt, &opkt, sizeof(AVPacket));
Debug(1, "writing video packet pts(%d) dts(%d) duration(%d)", opkt.pts, opkt.dts, opkt.duration );
if ((opkt.data == NULL)||(opkt.size < 1)) {
Warning("%s:%d: Mangled AVPacket: discarding frame", __FILE__, __LINE__ );
dumpPacket( ipkt);
dumpPacket(&opkt);
} else if ((video_previous_dts > 0) && (video_previous_dts > opkt.dts)) {
Warning("%s:%d: DTS out of order: %lld \u226E %lld; discarding frame", __FILE__, __LINE__, video_previous_dts, opkt.dts);
video_previous_dts = opkt.dts;
dumpPacket(&opkt);
} else {
video_previous_dts = opkt.dts; // Unsure if av_interleaved_write_frame() clobbers opkt.dts when out of order, so storing in advance
video_previous_pts = opkt.pts;
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);
}
}
zm_av_packet_unref(&opkt);
return 0;
} // end int VideoStore::writeVideoFramePacket( AVPacket *ipkt )
int VideoStore::writeAudioFramePacket( AVPacket *ipkt ) {
Debug(4, "writeAudioFrame");
if ( ! audio_output_stream ) {
Debug(1, "Called writeAudioFramePacket when no audio_output_stream");
return 0;//FIXME -ve return codes do not free packet in ffmpeg_camera at the moment
}
if ( audio_output_codec ) {
#ifdef HAVE_LIBAVRESAMPLE
#if LIBAVCODEC_VERSION_CHECK(57, 0, 0, 0, 0)
ret = avcodec_send_packet( audio_input_context, ipkt );
if ( ret < 0 ) {
Error("avcodec_send_packet fail %s", av_make_error_string(ret).c_str());
return 0;
}
ret = avcodec_receive_frame( audio_input_context, input_frame );
if ( ret < 0 ) {
Error("avcodec_receive_frame fail %s", av_make_error_string(ret).c_str());
return 0;
}
Debug(2, "Frame: samples(%d), format(%d), sample_rate(%d), channel layout(%d) refd(%d)",
input_frame->nb_samples,
input_frame->format,
input_frame->sample_rate,
input_frame->channel_layout,
audio_output_context->refcounted_frames
);
#else
/**
* Decode the audio frame stored in the packet.
* The input 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_input_context, input_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( &input_frame );
return 0;
}
if ( ! data_present ) {
Debug(2, "Not ready to transcode a frame yet.");
return 0;
}
#endif
int frame_size = input_frame->nb_samples;
Debug(4, "Frame size: %d", frame_size );
// Resample the input into the audioSampleBuffer until we proceed the whole decoded data
if ( (ret = avresample_convert( resample_context,
NULL,
0,
0,
input_frame->data,
0,
input_frame->nb_samples )) < 0 ) {
Error( "Could not resample frame (error '%s')\n",
av_make_error_string(ret).c_str());
return 0;
}
if ( avresample_available( resample_context ) < output_frame->nb_samples ) {
Debug(1, "No enough samples yet");
return 0;
}
// Read a frame audio data from the resample fifo
if ( avresample_read( resample_context, output_frame->data, output_frame->nb_samples ) != output_frame->nb_samples ) {
Warning( "Error reading resampled audio: " );
return 0;
}
av_init_packet(&opkt);
Debug(5, "after init packet" );
/** Set a timestamp based on the sample rate for the container. */
//output_frame->pts = av_rescale_q( opkt.pts, audio_output_context->time_base, audio_output_stream->time_base );
// convert the packet to the codec timebase from the stream timebase
//Debug(3, "output_frame->pts(%d) best effort(%d)", output_frame->pts,
//av_frame_get_best_effort_timestamp(output_frame)
//);
/**
* Encode the audio frame and store it in the temporary packet.
* The output audio stream encoder is used to do this.
*/
#if LIBAVCODEC_VERSION_CHECK(57, 0, 0, 0, 0)
if (( ret = avcodec_send_frame( audio_output_context, output_frame ) ) < 0 ) {
Error( "Could not send frame (error '%s')",
av_make_error_string(ret).c_str());
zm_av_packet_unref(&opkt);
return 0;
}
if (( ret = avcodec_receive_packet( audio_output_context, &opkt )) < 0 ) {
Error( "Could not recieve packet (error '%s')",
av_make_error_string(ret).c_str());
zm_av_packet_unref(&opkt);
return 0;
}
#else
if (( ret = avcodec_encode_audio2( audio_output_context, &opkt, output_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 output 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...
//Scale the PTS of the outgoing packet to be the correct time base
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 audeo_start_pts from (%d) to (%d)", audio_last_pts, ipkt->pts );
opkt.pts = audio_previous_pts + av_rescale_q(ipkt->pts, audio_input_stream->time_base, audio_output_stream->time_base);
} else {
opkt.pts = audio_previous_pts + av_rescale_q(ipkt->pts - audio_last_pts, audio_input_stream->time_base, audio_output_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;
}
//Scale the DTS of the outgoing packet to be the correct time base
if ( ! audio_last_dts ) {
opkt.dts = 0;
} else {
if( ipkt->dts == AV_NOPTS_VALUE ) {
// So if the input has no dts assigned... still need an output dts... so we use cur_dts?
if ( audio_last_dts > audio_input_stream->cur_dts ) {
Debug(1, "Resetting audio_last_dts from (%d) to cur_dts (%d)", audio_last_dts, audio_input_stream->cur_dts );
opkt.dts = audio_previous_dts + av_rescale_q( audio_input_stream->cur_dts, AV_TIME_BASE_Q, audio_output_stream->time_base);
} else {
opkt.dts = audio_previous_dts + av_rescale_q( audio_input_stream->cur_dts - audio_last_dts, AV_TIME_BASE_Q, audio_output_stream->time_base);
}
audio_last_dts = audio_input_stream->cur_dts;
Debug(2, "opkt.dts = %d from video_input_stream->cur_dts(%d) - last_dts(%d)", opkt.dts, audio_input_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_previous_dts + av_rescale_q(ipkt->dts, audio_input_stream->time_base, audio_output_stream->time_base);
} else {
opkt.dts = audio_previous_dts + av_rescale_q(ipkt->dts - audio_last_dts, audio_input_stream->time_base, audio_output_stream->time_base);
}
Debug(2, "opkt.dts = %d from ipkt->dts(%d) - last_dts(%d)", opkt.dts, ipkt->dts, audio_last_dts );
}
}
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?
opkt.duration = av_rescale_q(ipkt->duration, audio_input_stream->time_base, audio_output_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 = ipkt->stream_index;
Debug(2, "Stream index is %d", opkt.stream_index );
AVPacket safepkt;
memcpy(&safepkt, &opkt, sizeof(AVPacket));
audio_previous_dts = opkt.dts; // Unsure if av_interleaved_write_frame() clobbers opkt.dts when out of order, so storing in advance
audio_previous_pts = opkt.pts;
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 )
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