X-Git-Url: https://git.carlh.net/gitweb/?a=blobdiff_plain;f=src%2Flib%2Fffmpeg_decoder.cc;h=db88562ea09b2bb05001d14b2c031ad018c19f6f;hb=e29ce33a36c2e20444d57196defc86d5072bce81;hp=700e2983681750b17ecefcf07cd23a0037eac4ed;hpb=81646d1dd5b28fa05d8f134142dcbaed6314ebab;p=dcpomatic.git diff --git a/src/lib/ffmpeg_decoder.cc b/src/lib/ffmpeg_decoder.cc index 700e29836..db88562ea 100644 --- a/src/lib/ffmpeg_decoder.cc +++ b/src/lib/ffmpeg_decoder.cc @@ -114,22 +114,20 @@ FFmpegDecoder::flush () bool did_something = false; if (video) { - AVPacket packet; - av_init_packet (&packet); - packet.data = nullptr; - packet.size = 0; - if (decode_and_process_video_packet(&packet)) { + if (decode_and_process_video_packet(nullptr)) { did_something = true; } } for (auto i: ffmpeg_content()->ffmpeg_audio_streams()) { - AVPacket packet; - av_init_packet (&packet); - packet.data = nullptr; - packet.size = 0; - auto result = decode_audio_packet (i, &packet); - if (result.second) { + auto context = _codec_context[i->index(_format_context)]; + int r = avcodec_send_packet (context, nullptr); + if (r < 0 && r != AVERROR_EOF) { + /* EOF can happen if we've already sent a flush packet */ + throw DecodeError (N_("avcodec_send_packet"), N_("FFmpegDecoder::flush"), r); + } + r = avcodec_receive_frame (context, _frame); + if (r >= 0) { process_audio_frame (i); did_something = true; } @@ -224,33 +222,22 @@ FFmpegDecoder::pass () * Only the first buffer will be used for non-planar data, otherwise there will be one per channel. */ shared_ptr -FFmpegDecoder::deinterleave_audio (shared_ptr stream) const +FFmpegDecoder::deinterleave_audio (AVFrame* frame) { - DCPOMATIC_ASSERT (bytes_per_audio_sample (stream)); - - int const size = av_samples_get_buffer_size ( - 0, stream->stream(_format_context)->codecpar->channels, _frame->nb_samples, audio_sample_format (stream), 1 - ); - DCPOMATIC_ASSERT (size >= 0); + auto format = static_cast(frame->format); - /* XXX: can't we just use _frame->nb_samples directly here? */ /* XXX: can't we use swr_convert() to do the format conversion? */ - /* Deinterleave and convert to float */ - - /* total_samples and frames will be rounded down here, so if there are stray samples at the end - of the block that do not form a complete sample or frame they will be dropped. - */ - int const total_samples = size / bytes_per_audio_sample (stream); - int const channels = stream->channels(); - int const frames = total_samples / channels; + int const channels = frame->channels; + int const frames = frame->nb_samples; + int const total_samples = frames * channels; auto audio = make_shared(channels, frames); auto data = audio->data(); - switch (audio_sample_format (stream)) { + switch (format) { case AV_SAMPLE_FMT_U8: { - uint8_t* p = reinterpret_cast (_frame->data[0]); + auto p = reinterpret_cast (frame->data[0]); int sample = 0; int channel = 0; for (int i = 0; i < total_samples; ++i) { @@ -267,7 +254,7 @@ FFmpegDecoder::deinterleave_audio (shared_ptr stream) const case AV_SAMPLE_FMT_S16: { - int16_t* p = reinterpret_cast (_frame->data[0]); + auto p = reinterpret_cast (frame->data[0]); int sample = 0; int channel = 0; for (int i = 0; i < total_samples; ++i) { @@ -284,7 +271,7 @@ FFmpegDecoder::deinterleave_audio (shared_ptr stream) const case AV_SAMPLE_FMT_S16P: { - int16_t** p = reinterpret_cast (_frame->data); + auto p = reinterpret_cast (frame->data); for (int i = 0; i < channels; ++i) { for (int j = 0; j < frames; ++j) { data[i][j] = static_cast(p[i][j]) / (1 << 15); @@ -295,7 +282,7 @@ FFmpegDecoder::deinterleave_audio (shared_ptr stream) const case AV_SAMPLE_FMT_S32: { - int32_t* p = reinterpret_cast (_frame->data[0]); + auto p = reinterpret_cast (frame->data[0]); int sample = 0; int channel = 0; for (int i = 0; i < total_samples; ++i) { @@ -312,7 +299,7 @@ FFmpegDecoder::deinterleave_audio (shared_ptr stream) const case AV_SAMPLE_FMT_S32P: { - int32_t** p = reinterpret_cast (_frame->data); + auto p = reinterpret_cast (frame->data); for (int i = 0; i < channels; ++i) { for (int j = 0; j < frames; ++j) { data[i][j] = static_cast(p[i][j]) / 2147483648; @@ -323,7 +310,7 @@ FFmpegDecoder::deinterleave_audio (shared_ptr stream) const case AV_SAMPLE_FMT_FLT: { - float* p = reinterpret_cast (_frame->data[0]); + auto p = reinterpret_cast (frame->data[0]); int sample = 0; int channel = 0; for (int i = 0; i < total_samples; ++i) { @@ -340,20 +327,20 @@ FFmpegDecoder::deinterleave_audio (shared_ptr stream) const case AV_SAMPLE_FMT_FLTP: { - float** p = reinterpret_cast (_frame->data); - DCPOMATIC_ASSERT (_frame->channels <= channels); - /* Sometimes there aren't as many channels in the _frame as in the stream */ - for (int i = 0; i < _frame->channels; ++i) { + auto p = reinterpret_cast (frame->data); + DCPOMATIC_ASSERT (frame->channels <= channels); + /* Sometimes there aren't as many channels in the frame as in the stream */ + for (int i = 0; i < frame->channels; ++i) { memcpy (data[i], p[i], frames * sizeof(float)); } - for (int i = _frame->channels; i < channels; ++i) { + for (int i = frame->channels; i < channels; ++i) { audio->make_silent (i); } } break; default: - throw DecodeError (String::compose (_("Unrecognised audio sample format (%1)"), static_cast (audio_sample_format (stream)))); + throw DecodeError (String::compose(_("Unrecognised audio sample format (%1)"), static_cast(format))); } return audio; @@ -427,11 +414,9 @@ FFmpegDecoder::seek (ContentTime time, bool accurate) avcodec_flush_buffers (video_codec_context()); } -DCPOMATIC_DISABLE_WARNINGS for (auto i: ffmpeg_content()->ffmpeg_audio_streams()) { - avcodec_flush_buffers (i->stream(_format_context)->codec); + avcodec_flush_buffers (_codec_context[i->index(_format_context)]); } -DCPOMATIC_ENABLE_WARNINGS if (subtitle_codec_context ()) { avcodec_flush_buffers (subtitle_codec_context ()); @@ -466,7 +451,7 @@ FFmpegDecoder::audio_stream_from_index (int index) const void FFmpegDecoder::process_audio_frame (shared_ptr stream) { - auto data = deinterleave_audio (stream); + auto data = deinterleave_audio (_frame); ContentTime ct; if (_frame->pts == AV_NOPTS_VALUE) { @@ -513,27 +498,6 @@ FFmpegDecoder::process_audio_frame (shared_ptr stream) } -pair -FFmpegDecoder::decode_audio_packet (shared_ptr stream, AVPacket* packet) -{ - int frame_finished; - DCPOMATIC_DISABLE_WARNINGS - int decode_result = avcodec_decode_audio4 (stream->stream(_format_context)->codec, _frame, &frame_finished, packet); - DCPOMATIC_ENABLE_WARNINGS - if (decode_result < 0) { - /* avcodec_decode_audio4 can sometimes return an error even though it has decoded - some valid data; for example dca_subframe_footer can return AVERROR_INVALIDDATA - if it overreads the auxiliary data. ffplay carries on if frame_finished is true, - even in the face of such an error, so I think we should too. - - Returning from the method here caused mantis #352. - */ - LOG_WARNING ("avcodec_decode_audio4 failed (%1)", decode_result); - } - return make_pair(decode_result, frame_finished); -} - - void FFmpegDecoder::decode_and_process_audio_packet (AVPacket* packet) { @@ -542,33 +506,28 @@ FFmpegDecoder::decode_and_process_audio_packet (AVPacket* packet) return; } - /* Audio packets can contain multiple frames, so we may have to call avcodec_decode_audio4 - several times. Make a simple copy so we can alter data and size. - */ - AVPacket copy_packet = *packet; + auto context = _codec_context[stream->index(_format_context)]; - while (copy_packet.size > 0) { - auto result = decode_audio_packet (stream, ©_packet); - if (result.first < 0) { - /* avcodec_decode_audio4 can sometimes return an error even though it has decoded - some valid data; for example dca_subframe_footer can return AVERROR_INVALIDDATA - if it overreads the auxiliary data. ffplay carries on if frame_finished is true, - even in the face of such an error, so I think we should too. - - Returning from the method here caused mantis #352. - */ - } + int r = avcodec_send_packet (context, packet); + if (r < 0) { + /* We could cope with AVERROR(EAGAIN) and re-send the packet but I think it should never happen. + * Likewise I think AVERROR_EOF should not happen. + */ + throw DecodeError (N_("avcodec_send_packet"), N_("FFmpegDecoder::decode_and_process_audio_packet"), r); + } - if (result.second) { - process_audio_frame (stream); + while (r >= 0) { + r = avcodec_receive_frame (context, _frame); + if (r == AVERROR(EAGAIN)) { + /* More input is required */ + return; } - if (result.first) { - break; - } - - copy_packet.data += result.first; - copy_packet.size -= result.first; + /* We choose to be relaxed here about other errors; it seems that there may be valid + * data to decode even if an error occurred. #352 may be related (though this was + * when we were using an old version of the FFmpeg API). + */ + process_audio_frame (stream); } } @@ -578,12 +537,23 @@ FFmpegDecoder::decode_and_process_video_packet (AVPacket* packet) { DCPOMATIC_ASSERT (_video_stream); - int frame_finished; -DCPOMATIC_DISABLE_WARNINGS - if (avcodec_decode_video2 (video_codec_context(), _frame, &frame_finished, packet) < 0 || !frame_finished) { + auto context = video_codec_context(); + + int r = avcodec_send_packet (context, packet); + if (r < 0 && !(r == AVERROR_EOF && !packet)) { + /* We could cope with AVERROR(EAGAIN) and re-send the packet but I think it should never happen. + * AVERROR_EOF can happen during flush if we've already sent a flush packet. + */ + throw DecodeError (N_("avcodec_send_packet"), N_("FFmpegDecoder::decode_and_process_video_packet"), r); + } + + r = avcodec_receive_frame (context, _frame); + if (r == AVERROR(EAGAIN) || r == AVERROR_EOF) { + /* More input is required, or no more frames are coming */ return false; } -DCPOMATIC_ENABLE_WARNINGS + + /* We assume we'll only get one frame here, which I think is safe */ boost::mutex::scoped_lock lm (_filter_graphs_mutex);