mirror of
https://github.com/jojo61/vdr-plugin-softhdcuvid.git
synced 2025-12-27 07:21:07 +01:00
improve records cutting
This commit is contained in:
jojo61
227
codec.c
227
codec.c
@@ -490,8 +490,9 @@ next_part:
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if (ret >= 0) { // one is avail.
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got_frame = 1;
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}
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else
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else {
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got_frame = 0;
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}
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if (got_frame) { // frame completed
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#ifdef FFMPEG_WORKAROUND_ARTIFACTS
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@@ -1350,75 +1351,38 @@ void CodecAudioEnqueue(AudioDecoder * audio_decoder, int16_t * data, int count)
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int myavcodec_decode_audio3(AVCodecContext *avctx, int16_t *samples,
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int *frame_size_ptr,
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AVPacket *avpkt)
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AVPacket *avpkt)
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{
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AVFrame *frame = av_frame_alloc();
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int ret, got_frame = 0;
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AVFrame *frame = av_frame_alloc();
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int ret, got_frame = 0;
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if (!frame)
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return AVERROR(ENOMEM);
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#if 0
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if (avctx->get_buffer != avcodec_default_get_buffer) {
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av_log(avctx, AV_LOG_ERROR, "Custom get_buffer() for use with"
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"avcodec_decode_audio3() detected.ar *) samples Overriding with avcodec_default_get_buffer\n");
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av_log(avctx, AV_LOG_ERROR, "Please port your application to "
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"avcodec_decode_audio4()\n");
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avctx->get_buffer = avcodec_default_get_buffer;
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avctx->release_buffer = avcodec_default_release_buffer;
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}
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#endif
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ret = avcodec_decode_audio4(avctx, frame, &got_frame, avpkt);
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if (!frame)
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return AVERROR(ENOMEM);
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ret = avcodec_decode_audio4(avctx, frame, &got_frame, avpkt);
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if (ret >= 0 && got_frame) {
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#if 0
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int ch, plane_size;
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int planar = av_sample_fmt_is_planar(avctx->sample_fmt);
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int data_size = av_samples_get_buffer_size(&plane_size, avctx->channels,
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frame->nb_samples,
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avctx->sample_fmt, 1);
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if (*frame_size_ptr < data_size) {
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Debug(3, "output buffer size is too small for "
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"the current frame (%d < %d)\n", *frame_size_ptr, data_size);
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av_frame_free(&frame);
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return AVERROR(EINVAL);
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}
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#endif
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#if 1
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int i,ch;
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int planar = av_sample_fmt_is_planar(avctx->sample_fmt);
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int data_size = av_get_bytes_per_sample(avctx->sample_fmt);
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if (data_size < 0) {
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/* This should not occur, checking just for paranoia */
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fprintf(stderr, "Failed to calculate data size\n");
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exit(1);
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}
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for (i=0; i<frame->nb_samples; i++)
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for (ch=0; ch < avctx->channels; ch++) {
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memcpy(samples,frame->extended_data[ch]+data_size*i,data_size);
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samples = (char *) samples + data_size;
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if (ret >= 0 && got_frame) {
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int i,ch;
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int planar = av_sample_fmt_is_planar(avctx->sample_fmt);
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int data_size = av_get_bytes_per_sample(avctx->sample_fmt);
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if (data_size < 0) {
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/* This should not occur, checking just for paranoia */
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fprintf(stderr, "Failed to calculate data size\n");
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exit(1);
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}
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#endif
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//Debug(3,"data_size %d nb_samples %d sample_fmt %d channels %d planar %d\n",data_size,frame->nb_samples,avctx->sample_fmt,avctx->channels,planar);
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#if 0
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memcpy(samples, frame->extended_data[0], plane_size);
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// memcpy(samples, frame->data[0], plane_size);
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if (planar && avctx->channels > 1) {
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uint8_t *out = ((uint8_t *)samples) + plane_size;
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for (ch = 1; ch < avctx->channels; ch++) {
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memcpy(out, frame->extended_data[ch], plane_size);
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// memcpy(out, frame->data[ch], plane_size);
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out += plane_size;
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}
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}
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#endif
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*frame_size_ptr = data_size * avctx->channels * frame->nb_samples;
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} else {
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*frame_size_ptr = 0;
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}
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av_frame_free(&frame);
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return ret;
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for (i=0; i<frame->nb_samples; i++) {
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for (ch=0; ch < avctx->channels; ch++) {
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memcpy(samples,frame->extended_data[ch]+data_size*i,data_size);
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samples = (char *) samples + data_size;
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}
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}
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//Debug(3,"data_size %d nb_samples %d sample_fmt %d channels %d planar %d\n",data_size,frame->nb_samples,avctx->sample_fmt,avctx->channels,planar);
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*frame_size_ptr = data_size * avctx->channels * frame->nb_samples;
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} else {
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*frame_size_ptr = 0;
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}
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av_frame_free(&frame);
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return ret;
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}
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@@ -1444,110 +1408,59 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
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buf_sz = sizeof(buf);
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l = myavcodec_decode_audio3(audio_ctx, buf, &buf_sz, (AVPacket *) avpkt);
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if (avpkt->size != l) {
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if (l == AVERROR(EAGAIN)) {
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Error(_("codec: latm\n"));
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return;
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}
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if (l < 0) { // no audio frame could be decompressed
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Error(_("codec: error audio data\n"));
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return;
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}
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Error(_("codec: error more than one frame data\n"));
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if (l == AVERROR(EAGAIN)) {
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Error(_("codec: latm\n"));
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return;
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}
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if (l < 0) { // no audio frame could be decompressed
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Error(_("codec: error audio data\n"));
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return;
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}
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Error(_("codec: error more than one frame data\n"));
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}
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// update audio clock
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if (avpkt->pts != (int64_t) AV_NOPTS_VALUE) {
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CodecAudioSetClock(audio_decoder, avpkt->pts);
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CodecAudioSetClock(audio_decoder, avpkt->pts);
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}
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// FIXME: must first play remainings bytes, than change and play new.
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if (audio_decoder->Passthrough != CodecPassthrough
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|| audio_decoder->SampleRate != audio_ctx->sample_rate
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|| audio_decoder->Channels != audio_ctx->channels) {
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CodecAudioUpdateFormat(audio_decoder);
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|| audio_decoder->SampleRate != audio_ctx->sample_rate
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|| audio_decoder->Channels != audio_ctx->channels) {
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CodecAudioUpdateFormat(audio_decoder);
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}
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if (audio_decoder->HwSampleRate && audio_decoder->HwChannels) {
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// need to resample audio
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if (audio_decoder->ReSample) {
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int16_t outbuf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 4 +
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FF_INPUT_BUFFER_PADDING_SIZE]
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__attribute__ ((aligned(16)));
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int outlen;
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// need to resample audio
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if (audio_decoder->ReSample) {
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int16_t outbuf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 4 +
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FF_INPUT_BUFFER_PADDING_SIZE]
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__attribute__ ((aligned(16)));
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int outlen;
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// FIXME: libav-0.7.2 crash here
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outlen =
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audio_resample(audio_decoder->ReSample, outbuf, buf, buf_sz);
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// FIXME: libav-0.7.2 crash here
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outlen = audio_resample(audio_decoder->ReSample, outbuf, buf, buf_sz);
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#ifdef DEBUG
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if (outlen != buf_sz) {
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Debug(3, "codec/audio: possible fixed ffmpeg\n");
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}
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if (outlen != buf_sz) {
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Debug(3, "codec/audio: possible fixed ffmpeg\n");
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}
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#endif
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if (outlen) {
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// outlen seems to be wrong in ffmpeg-0.9
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outlen /= audio_decoder->Channels *
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av_get_bytes_per_sample(audio_ctx->sample_fmt);
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outlen *=
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audio_decoder->HwChannels *
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av_get_bytes_per_sample(audio_ctx->sample_fmt);
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Debug(4, "codec/audio: %d -> %d\n", buf_sz, outlen);
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CodecAudioEnqueue(audio_decoder, outbuf, outlen);
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}
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} else {
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if (CodecAudioPassthroughHelper(audio_decoder, avpkt)) {
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return;
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}
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#if 0
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//
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// old experimental code
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//
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if (1) {
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// FIXME: need to detect dts
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// copy original data for output
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// FIXME: buf is sint
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buf[0] = 0x72;
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buf[1] = 0xF8;
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buf[2] = 0x1F;
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buf[3] = 0x4E;
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buf[4] = 0x00;
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switch (avpkt->size) {
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case 512:
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buf[5] = 0x0B;
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break;
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case 1024:
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buf[5] = 0x0C;
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break;
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case 2048:
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buf[5] = 0x0D;
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break;
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default:
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Debug(3,
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"codec/audio: dts sample burst not supported\n");
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buf[5] = 0x00;
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break;
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if (outlen) {
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// outlen seems to be wrong in ffmpeg-0.9
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outlen /= audio_decoder->Channels *
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av_get_bytes_per_sample(audio_ctx->sample_fmt);
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outlen *=
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audio_decoder->HwChannels *
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av_get_bytes_per_sample(audio_ctx->sample_fmt);
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Debug(4, "codec/audio: %d -> %d\n", buf_sz, outlen);
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CodecAudioEnqueue(audio_decoder, outbuf, outlen);
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}
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} else {
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if (CodecAudioPassthroughHelper(audio_decoder, avpkt)) {
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return;
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}
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CodecAudioEnqueue(audio_decoder, buf, buf_sz);
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}
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buf[6] = (avpkt->size * 8);
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buf[7] = (avpkt->size * 8) >> 8;
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//buf[8] = 0x0B;
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//buf[9] = 0x77;
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//printf("%x %x\n", avpkt->data[0],avpkt->data[1]);
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// swab?
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memcpy(buf + 8, avpkt->data, avpkt->size);
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memset(buf + 8 + avpkt->size, 0, buf_sz - 8 - avpkt->size);
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} else if (1) {
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// FIXME: need to detect mp2
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// FIXME: mp2 passthrough
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// see softhddev.c version/layer
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// 0x04 mpeg1 layer1
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// 0x05 mpeg1 layer23
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// 0x06 mpeg2 ext
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// 0x07 mpeg2.5 layer 1
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// 0x08 mpeg2.5 layer 2
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// 0x09 mpeg2.5 layer 3
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}
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// DTS HD?
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// True HD?
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#endif
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CodecAudioEnqueue(audio_decoder, buf, buf_sz);
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}
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}
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}
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