mirror of
https://github.com/ZoneMinder/zoneminder.git
synced 2026-07-11 21:56:19 -04:00
fixes
This commit is contained in:
Isaac Connor
@@ -85,7 +85,6 @@ VideoStore::VideoStore(const char *filename_in, const char *format_in,
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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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Debug(2, "Success after metadata");
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output_format = oc->oformat;
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@@ -98,7 +97,16 @@ Debug(2, "Success after metadata");
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video_output_context = video_output_stream->codec;
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#if 0
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#if LIBAVCODEC_VERSION_CHECK(57, 0, 0, 0, 0)
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Debug(2, "setting parameters");
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ret = avcodec_parameters_to_context( video_output_context, video_input_stream->codecpar );
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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 getting parameters");
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}
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#else
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ret = avcodec_copy_context(video_output_context, video_input_context );
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if (ret < 0) {
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Fatal("Unable to copy input video context to output video context %s\n",
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@@ -106,31 +114,9 @@ Debug(2, "Success after metadata");
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} else {
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Debug(3, "Success copying context" );
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}
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#else
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#if 0
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Debug(2, "getting parameters");
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ret = avcodec_parameters_from_context( video_output_stream->codecpar, video_output_context );
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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 getting parameters");
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}
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#endif
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Debug(2, "setting parameters");
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ret = avcodec_parameters_to_context( video_output_context, video_input_stream->codecpar );
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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 getting parameters");
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}
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#endif
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Debug(3, "Time bases input stream time base(%d/%d) input codec tb: (%d/%d) video_output_stream->time-base(%d/%d) output codec tb (%d/%d)",
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Debug(3, "Time bases: VIDEO input stream (%d/%d) input codec: (%d/%d) output stream: (%d/%d) output codec (%d/%d)",
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video_input_stream->time_base.num,
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video_input_stream->time_base.den,
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video_input_context->time_base.num,
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@@ -188,16 +174,15 @@ Debug(2, "Success getting parameters");
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// WHY?
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//video_output_context->codec_tag = 0;
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if (!video_output_context->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, video_input_context->codec_tag) == video_output_context->codec_id
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|| av_codec_get_tag(oc->oformat->codec_tag, video_input_context->codec_id) <= 0) {
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Warning("Setting codec tag");
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video_output_context->codec_tag = video_input_context->codec_tag;
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}
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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, video_input_context->codec_tag) == video_output_context->codec_id
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|| av_codec_get_tag(oc->oformat->codec_tag, video_input_context->codec_id) <= 0) {
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Warning("Setting codec tag");
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video_output_context->codec_tag = video_input_context->codec_tag;
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}
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}
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if (oc->oformat->flags & AVFMT_GLOBALHEADER) {
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video_output_context->flags |= CODEC_FLAG_GLOBAL_HEADER;
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}
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@@ -226,7 +211,8 @@ Debug(2, "No codec_tag");
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audio_input_context = audio_input_stream->codec;
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if ( audio_input_context->codec_id != AV_CODEC_ID_AAC ) {
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Debug(3, "Got something other than AAC (%d)", audio_input_context->codec_id );
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avcodec_string(error_buffer, sizeof(error_buffer), audio_input_context, 0 );
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Debug(3, "Got something other than AAC (%s)", error_buffer );
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audio_output_stream = NULL;
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audio_output_codec = avcodec_find_encoder(AV_CODEC_ID_AAC);
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@@ -236,7 +222,6 @@ Debug(2, "Have audio output codec");
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audio_output_context = audio_output_stream->codec;
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//audio_output_context = avcodec_alloc_context3( audio_output_codec );
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if ( audio_output_context ) {
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Debug(2, "Have audio_output_context");
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@@ -267,32 +252,32 @@ Debug(2, "Have audio_output_context");
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}
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}
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/* check that the encoder supports s16 pcm input */
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if (!check_sample_fmt( audio_output_codec, audio_output_context->sample_fmt)) {
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Error( "Encoder does not support sample format %s, setting to FLTP",
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av_get_sample_fmt_name( audio_output_context->sample_fmt));
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audio_output_context->sample_fmt = AV_SAMPLE_FMT_FLTP;
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}
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Debug(3, "Audio Time bases input stream time base(%d/%d) input codec tb: (%d/%d) video_output_stream->time-base(%d/%d) output codec tb (%d/%d)",
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audio_input_stream->time_base.num,
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audio_input_stream->time_base.den,
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audio_input_context->time_base.num,
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audio_input_context->time_base.den,
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audio_output_stream->time_base.num,
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audio_output_stream->time_base.den,
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audio_output_context->time_base.num,
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audio_output_context->time_base.den
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);
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/** Set the sample rate for the container. */
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//audio_output_stream->time_base.den = audio_input_context->sample_rate;
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//audio_output_stream->time_base.num = 1;
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/* check that the encoder supports s16 pcm input */
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if (!check_sample_fmt( audio_output_codec, audio_output_context->sample_fmt)) {
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Error( "Encoder does not support sample format %s, setting to FLTP",
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av_get_sample_fmt_name( audio_output_context->sample_fmt));
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audio_output_context->sample_fmt = AV_SAMPLE_FMT_FLTP;
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}
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ret = avcodec_open2(audio_output_context, audio_output_codec, &opts );
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if ( ret < 0 ) {
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av_strerror(ret, error_buffer, sizeof(error_buffer));
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Fatal( "could not open codec (%d) (%s)\n", ret, error_buffer );
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} else {
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Debug(3, "Audio Time bases input stream (%d/%d) input codec: (%d/%d) output_stream (%d/%d) output codec (%d/%d)",
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audio_input_stream->time_base.num,
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audio_input_stream->time_base.den,
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audio_input_context->time_base.num,
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audio_input_context->time_base.den,
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audio_output_stream->time_base.num,
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audio_output_stream->time_base.den,
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audio_output_context->time_base.num,
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audio_output_context->time_base.den
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);
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/** Set the sample rate for the container. */
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//audio_output_stream->time_base.den = audio_input_context->sample_rate;
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//audio_output_stream->time_base.num = 1;
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ret = avcodec_open2(audio_output_context, audio_output_codec, &opts );
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if ( ret < 0 ) {
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av_strerror(ret, error_buffer, sizeof(error_buffer));
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Fatal( "could not open codec (%d) (%s)\n", ret, error_buffer );
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} else {
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Debug(1, "Audio output bit_rate (%d) sample_rate(%d) channels(%d) fmt(%d) layout(%d) frame_size(%d), refcounted_frames(%d)",
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audio_output_context->bit_rate,
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@@ -304,27 +289,27 @@ Debug(2, "Have audio_output_context");
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audio_output_context->refcounted_frames
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);
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#if 1
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/** Create the FIFO buffer based on the specified output sample format. */
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if (!(fifo = av_audio_fifo_alloc(audio_output_context->sample_fmt,
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audio_output_context->channels, 1))) {
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Error("Could not allocate FIFO\n");
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return;
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}
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/** Create the FIFO buffer based on the specified output sample format. */
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if (!(fifo = av_audio_fifo_alloc(audio_output_context->sample_fmt,
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audio_output_context->channels, 1))) {
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Error("Could not allocate FIFO\n");
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return;
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}
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#endif
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output_frame_size = audio_output_context->frame_size;
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/** Create a new frame to store the audio samples. */
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if (!(input_frame = zm_av_frame_alloc())) {
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Error("Could not allocate input frame");
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return;
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}
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/** Create a new frame to store the audio samples. */
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if (!(output_frame = zm_av_frame_alloc())) {
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Error("Could not allocate output frame");
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av_frame_free(&input_frame);
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return;
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}
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/**
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output_frame_size = audio_output_context->frame_size;
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/** Create a new frame to store the audio samples. */
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if (!(input_frame = zm_av_frame_alloc())) {
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Error("Could not allocate input frame");
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return;
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}
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/** Create a new frame to store the audio samples. */
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if (!(output_frame = zm_av_frame_alloc())) {
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Error("Could not allocate output frame");
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av_frame_free(&input_frame);
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return;
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}
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/**
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* Create a resampler context for the conversion.
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* Set the conversion parameters.
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* Default channel layouts based on the number of channels
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@@ -355,38 +340,38 @@ Debug(2, "Have audio_output_context");
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swr_free(&resample_context);
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return;
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}
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/**
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* Allocate as many pointers as there are audio channels.
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* Each pointer will later point to the audio samples of the corresponding
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* channels (although it may be NULL for interleaved formats).
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*/
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if (!( converted_input_samples = (uint8_t *)calloc( audio_output_context->channels, sizeof(*converted_input_samples))) ) {
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Error( "Could not allocate converted input sample pointers\n");
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return;
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}
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/**
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* Allocate memory for the samples of all channels in one consecutive
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* block for convenience.
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*/
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if ((ret = av_samples_alloc( &converted_input_samples, NULL,
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audio_output_context->channels,
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audio_output_context->frame_size,
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audio_output_context->sample_fmt, 0)) < 0) {
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Error( "Could not allocate converted input samples (error '%s')\n",
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av_make_error_string(ret).c_str() );
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av_freep(converted_input_samples);
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free(converted_input_samples);
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return;
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}
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Debug(2, "Success opening AAC codec");
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}
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av_dict_free(&opts);
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/**
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* Allocate as many pointers as there are audio channels.
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* Each pointer will later point to the audio samples of the corresponding
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* channels (although it may be NULL for interleaved formats).
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*/
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if (!( converted_input_samples = (uint8_t *)calloc( audio_output_context->channels, sizeof(*converted_input_samples))) ) {
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Error( "Could not allocate converted input sample pointers\n");
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return;
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}
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/**
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* Allocate memory for the samples of all channels in one consecutive
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* block for convenience.
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*/
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if ((ret = av_samples_alloc( &converted_input_samples, NULL,
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audio_output_context->channels,
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audio_output_context->frame_size,
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audio_output_context->sample_fmt, 0)) < 0) {
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Error( "Could not allocate converted input samples (error '%s')\n",
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av_make_error_string(ret).c_str() );
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av_freep(converted_input_samples);
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free(converted_input_samples);
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return;
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}
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Debug(2, "Success opening AAC codec");
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}
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av_dict_free(&opts);
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} else {
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Error( "could not allocate codec context for AAC\n");
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}
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} else {
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Error( "could not find codec for AAC\n");
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Error( "could not find codec for AAC\n");
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}
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} else {
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@@ -410,10 +395,10 @@ av_make_error_string(ret).c_str() );
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} else {
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Debug(3, "Audio is mono");
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}
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} // end if is AAC
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if (oc->oformat->flags & AVFMT_GLOBALHEADER) {
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audio_output_context->flags |= CODEC_FLAG_GLOBAL_HEADER;
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}
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} // end if is AAC
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Debug(3, "Audio Time bases input stream time base(%d/%d) input codec tb: (%d/%d) video_output_stream->time-base(%d/%d) output codec tb (%d/%d)",
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audio_input_stream->time_base.num,
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audio_input_stream->time_base.den,
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@@ -530,7 +515,7 @@ int VideoStore::writeVideoFramePacket( AVPacket *ipkt ) {
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if ( 1 ) {
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//Scale the PTS of the outgoing packet to be the correct time base
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if (ipkt->pts != AV_NOPTS_VALUE) {
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if ( video_start_pts < ipkt->pts ) {
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if ( (!video_start_pts) || (video_start_pts > ipkt->pts) ) {
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Debug(1, "Resetting video_start_pts from (%d) to (%d)", video_start_pts, ipkt->pts );
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//never gets set, so the first packet can set it.
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video_start_pts = ipkt->pts;
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@@ -546,7 +531,7 @@ if ( 1 ) {
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//Scale the DTS of the outgoing packet to be the correct time base
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if(ipkt->dts == AV_NOPTS_VALUE) {
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// why are we using cur_dts instead of packet.dts?
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if ( video_start_dts < video_input_stream->cur_dts ) {
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if ( (!video_start_dts) || (video_start_dts > video_input_stream->cur_dts) ) {
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Debug(1, "Resetting video_start_dts from (%d) to (%d) p.dts was (%d)", video_start_dts, video_input_stream->cur_dts, ipkt->dts );
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video_start_dts = video_input_stream->cur_dts;
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}
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@@ -555,9 +540,9 @@ if ( 1 ) {
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opkt.dts, video_input_stream->cur_dts, video_start_dts
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);
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} else {
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if ( video_start_dts < ipkt->dts ) {
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if ( (!video_start_dts) || (video_start_dts > ipkt->dts) ) {
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Debug(1, "Resetting video_start_dts from (%d) to (%d)", video_start_dts, ipkt->dts );
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video_start_dts = video_input_stream->cur_dts;
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video_start_dts = ipkt->dts;
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}
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opkt.dts = av_rescale_q(ipkt->dts - video_start_dts, video_input_stream->time_base, video_output_stream->time_base);
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Debug(3, "opkt.dts = %d from ipkt->dts(%d) - startDts(%d)", opkt.dts, ipkt->dts, video_start_dts );
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@@ -569,8 +554,19 @@ if ( 1 ) {
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opkt.duration = av_rescale_q(ipkt->duration, video_input_stream->time_base, video_output_stream->time_base);
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} else {
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// Using this results in super fast video output, might be because it should be using the codec time base instead of stream tb
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av_packet_rescale_ts( &opkt, video_input_stream->time_base, video_output_stream->time_base );
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}
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if ( opkt.dts != AV_NOPTS_VALUE ) {
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int64_t max = audio_output_stream->cur_dts + !(oc->oformat->flags & AVFMT_TS_NONSTRICT);
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if (audio_output_stream->cur_dts && audio_output_stream->cur_dts != AV_NOPTS_VALUE && max > opkt.dts) {
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Warning("st:%d PTS: %"PRId64" DTS: %"PRId64" < %"PRId64" invalid, clipping\n", opkt.stream_index, opkt.pts, opkt.dts, max);
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if( opkt.pts >= opkt.dts)
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opkt.pts = FFMAX(opkt.pts, max);
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opkt.dts = max;
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}
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}
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opkt.flags = ipkt->flags;
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opkt.pos=-1;
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@@ -650,9 +646,10 @@ int VideoStore::writeAudioFramePacket( AVPacket *ipkt ) {
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av_init_packet(&opkt);
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Debug(5, "after init packet" );
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#if 1
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//Scale the PTS of the outgoing packet to be the correct time base
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if (ipkt->pts != AV_NOPTS_VALUE) {
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if ( audio_start_pts < ipkt->pts ) {
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if ( (!audio_start_pts) || ( audio_start_pts > ipkt->pts ) ) {
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Debug(1, "Resetting audeo_start_pts from (%d) to (%d)", audio_start_pts, ipkt->pts );
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//never gets set, so the first packet can set it.
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audio_start_pts = ipkt->pts;
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@@ -665,8 +662,8 @@ int VideoStore::writeAudioFramePacket( AVPacket *ipkt ) {
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//Scale the DTS of the outgoing packet to be the correct time base
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if(ipkt->dts == AV_NOPTS_VALUE) {
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if ( audio_start_dts < audio_input_stream->cur_dts ) {
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Debug(1, "Resetting audeo_start_pts from (%d) to (%d)", audio_start_pts, audio_input_stream->cur_dts );
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if ( (!audio_start_dts) || (audio_start_dts > audio_input_stream->cur_dts ) ) {
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Debug(1, "Resetting audeo_start_pts from (%d) to (%d)", audio_start_dts, audio_input_stream->cur_dts );
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audio_start_dts = audio_input_stream->cur_dts;
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}
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opkt.dts = av_rescale_q(audio_input_stream->cur_dts - audio_start_dts, AV_TIME_BASE_Q, audio_output_stream->time_base);
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@@ -674,7 +671,10 @@ int VideoStore::writeAudioFramePacket( AVPacket *ipkt ) {
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opkt.dts, audio_input_stream->cur_dts, audio_start_dts
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);
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} else {
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if ( ! audio_start_dts ) audio_start_dts = ipkt->dts;
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if ( (!audio_start_dts) || ( audio_start_dts > ipkt->dts ) ) {
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Debug(1, "Resetting audeo_start_dts from (%d) to (%d)", audio_start_dts, ipkt->dts );
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audio_start_dts = ipkt->dts;
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}
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opkt.dts = av_rescale_q(ipkt->dts - audio_start_dts, audio_input_stream->time_base, audio_output_stream->time_base);
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Debug(2, "opkt.dts = %d from ipkt->dts(%d) - startDts(%d)", opkt.dts, ipkt->dts, audio_start_dts );
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}
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@@ -686,11 +686,14 @@ int VideoStore::writeAudioFramePacket( AVPacket *ipkt ) {
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//opkt.dts = AV_NOPTS_VALUE;
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opkt.duration = av_rescale_q(ipkt->duration, audio_input_stream->time_base, audio_output_stream->time_base);
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#else
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#endif
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// pkt.pos: byte position in stream, -1 if unknown
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opkt.pos = -1;
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opkt.flags = ipkt->flags;
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opkt.stream_index = ipkt->stream_index;
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Debug(3, "Stream index is %d", opkt.stream_index );
|
||||
Debug(2, "Stream index is %d", opkt.stream_index );
|
||||
|
||||
if ( audio_output_codec ) {
|
||||
|
||||
@@ -733,6 +736,8 @@ av_codec_is_encoder( audio_output_context->codec)
|
||||
av_frame_unref( input_frame );
|
||||
#else
|
||||
|
||||
// convert the packet to the codec timebase from the stream timebase
|
||||
av_packet_rescale_ts( ipkt, audio_input_stream->time_base, audio_input_context->time_base );
|
||||
|
||||
/**
|
||||
* Decode the audio frame stored in the packet.
|
||||
@@ -749,34 +754,38 @@ av_codec_is_encoder( audio_output_context->codec)
|
||||
zm_av_unref_packet(&opkt);
|
||||
return 0;
|
||||
}
|
||||
if ( data_present ) {
|
||||
if ( ! data_present ) {
|
||||
Debug(2, "Not ready to transcode a frame yet.");
|
||||
zm_av_unref_packet(&opkt);
|
||||
return 0;
|
||||
}
|
||||
|
||||
int frame_size = input_frame->nb_samples;
|
||||
Debug(2, "Frame size: %d", frame_size );
|
||||
int frame_size = input_frame->nb_samples;
|
||||
Debug(4, "Frame size: %d", frame_size );
|
||||
|
||||
|
||||
Debug(2, "About to convert");
|
||||
Debug(4, "About to convert");
|
||||
|
||||
/** Convert the samples using the resampler. */
|
||||
if ((ret = swr_convert(resample_context,
|
||||
&converted_input_samples, frame_size,
|
||||
(const uint8_t **)input_frame->extended_data , frame_size)) < 0) {
|
||||
Error( "Could not convert input samples (error '%s')\n",
|
||||
av_make_error_string(ret).c_str()
|
||||
);
|
||||
return 0;
|
||||
&converted_input_samples, frame_size,
|
||||
(const uint8_t **)input_frame->extended_data , frame_size)) < 0) {
|
||||
Error( "Could not convert input samples (error '%s')\n",
|
||||
av_make_error_string(ret).c_str()
|
||||
);
|
||||
return 0;
|
||||
}
|
||||
|
||||
Debug(2, "About to realloc");
|
||||
Debug(4, "About to realloc");
|
||||
if ((ret = av_audio_fifo_realloc(fifo, av_audio_fifo_size(fifo) + frame_size)) < 0) {
|
||||
Error( "Could not reallocate FIFO to %d\n", av_audio_fifo_size(fifo) + frame_size );
|
||||
return 0;
|
||||
Error( "Could not reallocate FIFO to %d\n", av_audio_fifo_size(fifo) + frame_size );
|
||||
return 0;
|
||||
}
|
||||
/** Store the new samples in the FIFO buffer. */
|
||||
Debug(2, "About to write");
|
||||
Debug(4, "About to write");
|
||||
if (av_audio_fifo_write(fifo, (void **)&converted_input_samples, frame_size) < frame_size) {
|
||||
Error( "Could not write data to FIFO\n");
|
||||
return 0;
|
||||
Error( "Could not write data to FIFO\n");
|
||||
return 0;
|
||||
}
|
||||
|
||||
/** Create a new frame to store the audio samples. */
|
||||
@@ -784,60 +793,62 @@ Debug(2, "About to write");
|
||||
Error("Could not allocate output frame");
|
||||
return 0;
|
||||
}
|
||||
/**
|
||||
* Set the frame's parameters, especially its size and format.
|
||||
* av_frame_get_buffer needs this to allocate memory for the
|
||||
* audio samples of the frame.
|
||||
* Default channel layouts based on the number of channels
|
||||
* are assumed for simplicity.
|
||||
*/
|
||||
output_frame->nb_samples = audio_output_context->frame_size;
|
||||
output_frame->channel_layout = audio_output_context->channel_layout;
|
||||
output_frame->channels = audio_output_context->channels;
|
||||
output_frame->format = audio_output_context->sample_fmt;
|
||||
output_frame->sample_rate = audio_output_context->sample_rate;
|
||||
/**
|
||||
* Allocate the samples of the created frame. This call will make
|
||||
* sure that the audio frame can hold as many samples as specified.
|
||||
*/
|
||||
Debug(2, "getting buffer");
|
||||
if (( ret = av_frame_get_buffer( output_frame, 0)) < 0) {
|
||||
Error( "Couldnt allocate output frame buffer samples (error '%s')",
|
||||
av_make_error_string(ret).c_str() );
|
||||
Error("Frame: samples(%d) layout (%d) format(%d) rate(%d)", output_frame->nb_samples,
|
||||
output_frame->channel_layout, output_frame->format , output_frame->sample_rate
|
||||
);
|
||||
zm_av_unref_packet(&opkt);
|
||||
return 0;
|
||||
}
|
||||
|
||||
/** Set a timestamp based on the sample rate for the container. */
|
||||
if (output_frame) {
|
||||
output_frame->pts = opkt.pts;
|
||||
}
|
||||
Debug(2, "About to read");
|
||||
if (av_audio_fifo_read(fifo, (void **)output_frame->data, frame_size) < frame_size) {
|
||||
Error( "Could not read data from FIFO\n");
|
||||
return 0;
|
||||
/**
|
||||
* Set the frame's parameters, especially its size and format.
|
||||
* av_frame_get_buffer needs this to allocate memory for the
|
||||
* audio samples of the frame.
|
||||
* Default channel layouts based on the number of channels
|
||||
* are assumed for simplicity.
|
||||
*/
|
||||
output_frame->nb_samples = audio_output_context->frame_size;
|
||||
output_frame->channel_layout = audio_output_context->channel_layout;
|
||||
output_frame->channels = audio_output_context->channels;
|
||||
output_frame->format = audio_output_context->sample_fmt;
|
||||
output_frame->sample_rate = audio_output_context->sample_rate;
|
||||
/**
|
||||
* Allocate the samples of the created frame. This call will make
|
||||
* sure that the audio frame can hold as many samples as specified.
|
||||
*/
|
||||
Debug(4, "getting buffer");
|
||||
if (( ret = av_frame_get_buffer( output_frame, 0)) < 0) {
|
||||
Error( "Couldnt allocate output frame buffer samples (error '%s')",
|
||||
av_make_error_string(ret).c_str() );
|
||||
Error("Frame: samples(%d) layout (%d) format(%d) rate(%d)", output_frame->nb_samples,
|
||||
output_frame->channel_layout, output_frame->format , output_frame->sample_rate
|
||||
);
|
||||
zm_av_unref_packet(&opkt);
|
||||
return 0;
|
||||
}
|
||||
/**
|
||||
* Encode the audio frame and store it in the temporary packet.
|
||||
* The output audio stream encoder is used to do this.
|
||||
*/
|
||||
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_unref_packet(&opkt);
|
||||
return 0;
|
||||
}
|
||||
|
||||
//av_frame_unref( output_frame);
|
||||
//av_frame_free( &output_frame );
|
||||
/** Set a timestamp based on the sample rate for the container. */
|
||||
if (output_frame) {
|
||||
output_frame->pts = av_frame_get_best_effort_timestamp(output_frame);
|
||||
}
|
||||
Debug(4, "About to read");
|
||||
if (av_audio_fifo_read(fifo, (void **)output_frame->data, frame_size) < frame_size) {
|
||||
Error( "Could not read data from FIFO\n");
|
||||
return 0;
|
||||
}
|
||||
/**
|
||||
* Encode the audio frame and store it in the temporary packet.
|
||||
* The output audio stream encoder is used to do this.
|
||||
*/
|
||||
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_unref_packet(&opkt);
|
||||
return 0;
|
||||
}
|
||||
if ( ! data_present ) {
|
||||
Debug(2, "Not ready to output a frame yet.");
|
||||
zm_av_unref_packet(&opkt);
|
||||
return 0;
|
||||
}
|
||||
|
||||
// Convert tb from code back to stream
|
||||
av_packet_rescale_ts(&opkt, audio_output_context->time_base, audio_output_stream->time_base);
|
||||
|
||||
} else {
|
||||
Debug(2, "Not data present" );
|
||||
} // end if data_present
|
||||
#endif
|
||||
} else {
|
||||
opkt.data = ipkt->data;
|
||||
@@ -849,12 +860,6 @@ Debug(2, "About to read");
|
||||
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());
|
||||
opkt.pts = 0;
|
||||
opkt.dts = 0;
|
||||
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" );
|
||||
|
||||
Reference in New Issue
Block a user