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This commit is contained in:
Isaac Connor
commit
9656032906
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@ -210,6 +210,9 @@ VideoStore::VideoStore(
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out_frame = NULL;
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#if defined(HAVE_LIBSWRESAMPLE) || defined(HAVE_LIBAVRESAMPLE)
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resample_ctx = NULL;
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#if defined(HAVE_LIBSWRESAMPLE)
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fifo = NULL;
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#endif
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#endif
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if ( audio_in_stream ) {
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@ -471,6 +474,10 @@ VideoStore::~VideoStore() {
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#if defined(HAVE_LIBAVRESAMPLE) || defined(HAVE_LIBSWRESAMPLE)
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if ( resample_ctx ) {
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#if defined(HAVE_LIBSWRESAMPLE)
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if ( fifo ) {
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av_audio_fifo_free(fifo);
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fifo = NULL;
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}
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swr_free(&resample_ctx);
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#else
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#if defined(HAVE_LIBAVRESAMPLE)
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@ -569,7 +576,7 @@ bool VideoStore::setup_resampler() {
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if ( audio_out_codec->supported_samplerates ) {
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int found = 0;
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for ( unsigned int i = 0; audio_out_codec->supported_samplerates[i]; i++) {
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for ( unsigned int i = 0; audio_out_codec->supported_samplerates[i]; i++ ) {
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if ( audio_out_ctx->sample_rate ==
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audio_out_codec->supported_samplerates[i] ) {
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found = 1;
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@ -602,7 +609,7 @@ bool VideoStore::setup_resampler() {
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ret = avcodec_open2(audio_out_ctx, audio_out_codec, &opts);
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av_dict_free(&opts);
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if ( ret < 0 ) {
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Error("could not open codec (%d) (%s)\n", ret, av_make_error_string(ret).c_str());
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Error("could not open codec (%d) (%s)", ret, av_make_error_string(ret).c_str());
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audio_out_codec = NULL;
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audio_out_ctx = NULL;
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audio_out_stream = NULL;
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@ -618,6 +625,12 @@ bool VideoStore::setup_resampler() {
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}
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#endif
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Debug(1,
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"Audio in bit_rate (%d) sample_rate(%d) channels(%d) fmt(%d) "
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"layout(%d) frame_size(%d)",
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audio_in_ctx->bit_rate, audio_in_ctx->sample_rate,
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audio_in_ctx->channels, audio_in_ctx->sample_fmt,
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audio_in_ctx->channel_layout, audio_in_ctx->frame_size);
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Debug(1,
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"Audio out bit_rate (%d) sample_rate(%d) channels(%d) fmt(%d) "
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"layout(%d) frame_size(%d)",
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@ -639,6 +652,12 @@ bool VideoStore::setup_resampler() {
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}
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#if defined(HAVE_LIBSWRESAMPLE)
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if (!(fifo = av_audio_fifo_alloc(
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audio_out_ctx->sample_fmt,
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audio_out_ctx->channels, 1))) {
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Error("Could not allocate FIFO");
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return false;
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}
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resample_ctx = swr_alloc_set_opts(NULL,
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audio_out_ctx->channel_layout,
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audio_out_ctx->sample_fmt,
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@ -725,7 +744,7 @@ bool VideoStore::setup_resampler() {
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out_frame, audio_out_ctx->channels,
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audio_out_ctx->sample_fmt,
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(const uint8_t *)converted_in_samples,
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audioSampleBuffer_size, 0) < 0) {
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audioSampleBuffer_size, 0) < 0 ) {
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Error("Could not allocate converted in sample pointers");
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return false;
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}
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@ -749,10 +768,8 @@ int VideoStore::writeVideoFramePacket(AVPacket *ipkt) {
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av_rescale_q(ipkt->pts - video_last_pts, video_in_stream->time_base,
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video_out_stream->time_base);
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Debug(1, "duration calc: pts(%" PRId64 ") - last_pts(% " PRId64 ") = (%" PRId64 ")",
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ipkt->pts,
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video_last_pts,
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duration);
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if (duration <= 0) {
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ipkt->pts, video_last_pts, duration);
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if ( duration <= 0 ) {
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duration = ipkt->duration ? ipkt->duration : av_rescale_q(1,video_in_stream->time_base, video_out_stream->time_base);
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}
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}
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@ -824,7 +841,7 @@ int VideoStore::writeVideoFramePacket(AVPacket *ipkt) {
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}
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}
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#endif
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if (opkt.dts > opkt.pts) {
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if ( opkt.dts > opkt.pts ) {
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Debug(1,
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"opkt.dts(%d) must be <= opkt.pts(%d). Decompression must happen "
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"before presentation.",
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@ -834,22 +851,20 @@ int VideoStore::writeVideoFramePacket(AVPacket *ipkt) {
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opkt.flags = ipkt->flags;
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opkt.pos = -1;
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opkt.data = ipkt->data;
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opkt.size = ipkt->size;
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opkt.stream_index = video_out_stream->index;
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AVPacket safepkt;
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memcpy(&safepkt, &opkt, sizeof(AVPacket));
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dumpPacket( &opkt, "writing video packet" );
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if ((opkt.data == NULL) || (opkt.size < 1)) {
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dumpPacket(&opkt, "writing video packet");
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if ( (opkt.data == NULL) || (opkt.size < 1) ) {
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Warning("%s:%d: Mangled AVPacket: discarding frame", __FILE__, __LINE__);
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dumpPacket(ipkt);
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dumpPacket(&opkt);
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} else if ((video_next_dts > 0) && (video_next_dts > opkt.dts)) {
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} else if ( (video_next_dts > 0) && (video_next_dts > opkt.dts) ) {
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Warning("%s:%d: DTS out of order: %lld \u226E %lld; discarding frame",
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__FILE__, __LINE__, video_next_dts, opkt.dts);
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video_next_dts = opkt.dts;
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@ -859,12 +874,11 @@ int VideoStore::writeVideoFramePacket(AVPacket *ipkt) {
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video_next_dts = opkt.dts + duration;
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video_next_pts = opkt.pts + duration;
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ret = av_interleaved_write_frame(oc, &opkt);
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if (ret < 0) {
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if ( ret < 0 ) {
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// There's nothing we can really do if the frame is rejected, just drop it
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// and get on with the next
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Warning(
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"%s:%d: Writing frame [av_interleaved_write_frame()] failed: %s(%d) "
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" ",
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"%s:%d: Writing frame [av_interleaved_write_frame()] failed: %s(%d)"
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__FILE__, __LINE__, av_make_error_string(ret).c_str(), ret);
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dumpPacket(&safepkt);
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#if LIBAVCODEC_VERSION_CHECK(57, 64, 0, 64, 0)
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@ -932,36 +946,63 @@ int VideoStore::writeAudioFramePacket(AVPacket *ipkt) {
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return 0;
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}
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#endif
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int frame_size = in_frame->nb_samples;
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// Resample the in into the audioSampleBuffer until we proceed the whole
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// decoded data
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zm_dump_frame(in_frame, "In frame");
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zm_dump_frame(out_frame, "Out frame before resample");
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Debug(2, "Converting %d to %d samples", in_frame->nb_samples, out_frame->nb_samples);
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#if defined(HAVE_LIBSWRESAMPLE)
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#if 0
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(ret = swr_convert(resample_ctx,
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out_frame->data, frame_size,
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(const uint8_t**)in_frame->data,
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in_frame->nb_samples
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))
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ret = swr_convert(resample_ctx,
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out_frame->data, frame_size,
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(const uint8_t**)in_frame->data,
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in_frame->nb_samples
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);
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#else
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ret = swr_convert_frame(resample_ctx, out_frame, in_frame);
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#endif
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#else
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#if defined(HAVE_LIBAVRESAMPLE)
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ret = avresample_convert(resample_ctx, NULL, 0, 0, in_frame->data,
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0, in_frame->nb_samples);
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#endif
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#endif
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out_frame->pts = in_frame->pts;
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av_frame_unref(in_frame);
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if ( ret < 0 ) {
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Error("Could not resample frame (error '%s')",
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av_make_error_string(ret).c_str());
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return 0;
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}
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#endif
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if ((ret = av_audio_fifo_realloc(fifo, av_audio_fifo_size(fifo) + out_frame->nb_samples)) < 0) {
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Error("Could not reallocate FIFO");
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return 0;
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}
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/** Store the new samples in the FIFO buffer. */
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ret = av_audio_fifo_write(fifo, (void **)out_frame->data, out_frame->nb_samples);
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if ( ret < frame_size ) {
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Error("Could not write data to FIFO on %d written", ret);
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return 0;
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}
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#if defined(HAVE_LIBAVRESAMPLE)
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// Reset frame_size to output_frame_size
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frame_size = audio_out_ctx->frame_size;
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// AAC requires 1024 samples per encode. Our input tends to be 160, so need to buffer them.
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if ( frame_size > av_audio_fifo_size(fifo) ) {
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return 0;
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}
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if ( av_audio_fifo_read(fifo, (void **)out_frame->data, frame_size) < frame_size ) {
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Error("Could not read data from FIFO");
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return 0;
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}
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out_frame->nb_samples = frame_size;
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/// FIXME this is not the correct pts
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out_frame->pts = in_frame->pts;
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#else
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#if defined(HAVE_LIBAVRESAMPLE)
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(ret = avresample_convert(resample_ctx, NULL, 0, 0, in_frame->data,
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0, in_frame->nb_samples))
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av_frame_unref(in_frame);
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if ( ret < 0 ) {
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Error("Could not resample frame (error '%s')",
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av_make_error_string(ret).c_str());
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return 0;
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}
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int samples_available = avresample_available(resample_ctx);
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if ( samples_available < frame_size ) {
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@ -975,11 +1016,11 @@ int VideoStore::writeAudioFramePacket(AVPacket *ipkt) {
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Warning("Error reading resampled audio:");
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return 0;
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}
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#endif
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#endif
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zm_dump_frame(out_frame,"Out frame after resample");
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zm_dump_frame(out_frame, "Out frame after resample");
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av_init_packet(&opkt);
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Debug(5, "after init packet");
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#if LIBAVCODEC_VERSION_CHECK(57, 64, 0, 64, 0)
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if ( (ret = avcodec_send_frame(audio_out_ctx, out_frame)) < 0 ) {
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@ -993,7 +1034,7 @@ int VideoStore::writeAudioFramePacket(AVPacket *ipkt) {
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if ( (ret = avcodec_receive_packet(audio_out_ctx, &opkt)) < 0 ) {
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if ( AVERROR(EAGAIN) == ret ) {
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// THe codec may need more samples than it has, perfectly valid
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// The codec may need more samples than it has, perfectly valid
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Debug(3, "Could not recieve packet (error '%s')",
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av_make_error_string(ret).c_str());
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} else {
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@ -1083,7 +1124,7 @@ int VideoStore::writeAudioFramePacket(AVPacket *ipkt) {
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}
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#endif
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// audio_last_dts = ipkt->dts;
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if (opkt.dts > opkt.pts) {
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if ( opkt.dts > opkt.pts ) {
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Debug(1,
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"opkt.dts(%d) must be <= opkt.pts(%d). Decompression must happen "
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"before presentation.",
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@ -1093,16 +1134,16 @@ int VideoStore::writeAudioFramePacket(AVPacket *ipkt) {
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// I wonder if we could just use duration instead of all the hoop jumping
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// above?
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//
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if (out_frame) {
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if ( out_frame ) {
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opkt.duration = out_frame->nb_samples;
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} else {
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opkt.duration = ipkt->duration;
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}
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// opkt.duration = av_rescale_q(ipkt->duration, audio_in_stream->time_base,
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// audio_out_stream->time_base);
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Debug(2, "opkt.pts (%d), opkt.dts(%d) opkt.duration = (%d)", opkt.pts,
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opkt.dts, opkt.duration);
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Debug(2, "opkt.pts (%d), opkt.dts(%d) opkt.duration = (%d)",
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opkt.pts, opkt.dts, opkt.duration);
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// pkt.pos: byte position in stream, -1 if unknown
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opkt.pos = -1;
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@ -1113,8 +1154,8 @@ int VideoStore::writeAudioFramePacket(AVPacket *ipkt) {
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AVPacket safepkt;
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memcpy(&safepkt, &opkt, sizeof(AVPacket));
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ret = av_interleaved_write_frame(oc, &opkt);
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if (ret != 0) {
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Error("Error writing audio frame packet: %s\n",
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if ( ret != 0 ) {
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Error("Error writing audio frame packet: %s",
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av_make_error_string(ret).c_str());
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dumpPacket(&safepkt);
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} else {
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@ -1122,4 +1163,4 @@ int VideoStore::writeAudioFramePacket(AVPacket *ipkt) {
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}
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zm_av_packet_unref(&opkt);
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return 0;
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} // end int VideoStore::writeAudioFramePacket( AVPacket *ipkt )
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} // end int VideoStore::writeAudioFramePacket(AVPacket *ipkt)
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@ -5,6 +5,7 @@
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extern "C" {
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#ifdef HAVE_LIBSWRESAMPLE
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#include "libswresample/swresample.h"
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#include "libavutil/audio_fifo.h"
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#else
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#ifdef HAVE_LIBAVRESAMPLE
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#include "libavresample/avresample.h"
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@ -44,6 +45,7 @@ private:
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AVCodecContext *audio_out_ctx;
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#ifdef HAVE_LIBSWRESAMPLE
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SwrContext *resample_ctx;
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AVAudioFifo *fifo;
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#else
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#ifdef HAVE_LIBAVRESAMPLE
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AVAudioResampleContext* resample_ctx;
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