#include <utility>
#include <vector>
+#include <Eigen/Core>
+#include <Eigen/LU>
+#include <movit/colorspace_conversion_effect.h>
+
#include "bmusb/bmusb.h"
#include "shared/ffmpeg_raii.h"
#include "ffmpeg_util.h"
using namespace std::chrono;
using namespace bmusb;
using namespace movit;
+using namespace Eigen;
namespace {
return av_get_pix_fmt(best_format);
}
-YCbCrFormat decode_ycbcr_format(const AVPixFmtDescriptor *desc, const AVFrame *frame)
+YCbCrFormat decode_ycbcr_format(const AVPixFmtDescriptor *desc, const AVFrame *frame, bool is_mjpeg)
{
YCbCrFormat format;
- AVColorSpace colorspace = av_frame_get_colorspace(frame);
+ AVColorSpace colorspace = frame->colorspace;
switch (colorspace) {
case AVCOL_SPC_BT709:
format.luma_coefficients = YCBCR_REC_709;
format.cb_y_position = 1.0;
break;
default:
- fprintf(stderr, "Unknown chroma location coefficient enum %d from FFmpeg; choosing Rec. 709.\n",
+ fprintf(stderr, "Unknown chroma location coefficient enum %d from FFmpeg; choosing center.\n",
frame->chroma_location);
format.cb_x_position = 0.5;
format.cb_y_position = 0.5;
break;
}
+ if (is_mjpeg && !format.full_range) {
+ // Limited-range MJPEG is only detected by FFmpeg whenever a special
+ // JPEG comment is set, which means that in practice, the stream is
+ // almost certainly generated by Futatabi. Override FFmpeg's forced
+ // MJPEG defaults (it disregards the values set in the mux) with what
+ // Futatabi sets.
+ format.luma_coefficients = YCBCR_REC_709;
+ format.cb_x_position = 0.0;
+ format.cb_y_position = 0.5;
+ }
+
format.cr_x_position = format.cb_x_position;
format.cr_y_position = format.cb_y_position;
return format;
}
+RGBTriplet get_neutral_color(AVDictionary *metadata)
+{
+ if (metadata == nullptr) {
+ return RGBTriplet(1.0f, 1.0f, 1.0f);
+ }
+ AVDictionaryEntry *entry = av_dict_get(metadata, "WhitePoint", nullptr, 0);
+ if (entry == nullptr) {
+ return RGBTriplet(1.0f, 1.0f, 1.0f);
+ }
+
+ unsigned x_nom, x_den, y_nom, y_den;
+ if (sscanf(entry->value, " %u:%u , %u:%u", &x_nom, &x_den, &y_nom, &y_den) != 4) {
+ fprintf(stderr, "WARNING: Unable to parse white point '%s', using default white point\n", entry->value);
+ return RGBTriplet(1.0f, 1.0f, 1.0f);
+ }
+
+ double x = double(x_nom) / x_den;
+ double y = double(y_nom) / y_den;
+ double z = 1.0 - x - y;
+
+ Matrix3d rgb_to_xyz_matrix = movit::ColorspaceConversionEffect::get_xyz_matrix(COLORSPACE_sRGB);
+ Vector3d rgb = rgb_to_xyz_matrix.inverse() * Vector3d(x, y, z);
+
+ return RGBTriplet(rgb[0], rgb[1], rgb[2]);
+}
+
} // namespace
FFmpegCapture::FFmpegCapture(const string &filename, unsigned width, unsigned height)
if (has_dequeue_callbacks) {
dequeue_cleanup_callback();
}
- avresample_free(&resampler);
+ swr_free(&resampler);
}
void FFmpegCapture::configure_card()
last_modified = buf.st_mtim;
}
- AVDictionary *opts = nullptr;
- av_dict_set(&opts, "fflags", "nobuffer", 0);
-
- auto format_ctx = avformat_open_input_unique(pathname.c_str(), nullptr, &opts, AVIOInterruptCB{ &FFmpegCapture::interrupt_cb_thunk, this });
+ auto format_ctx = avformat_open_input_unique(pathname.c_str(), nullptr, nullptr, AVIOInterruptCB{ &FFmpegCapture::interrupt_cb_thunk, this });
if (format_ctx == nullptr) {
fprintf(stderr, "%s: Error opening file\n", pathname.c_str());
return false;
}
int audio_stream_index = find_stream_index(format_ctx.get(), AVMEDIA_TYPE_AUDIO);
+ int subtitle_stream_index = find_stream_index(format_ctx.get(), AVMEDIA_TYPE_SUBTITLE);
+ has_last_subtitle = false;
// Open video decoder.
const AVCodecParameters *video_codecpar = format_ctx->streams[video_stream_index]->codecpar;
unique_ptr<AVCodecContext, decltype(avcodec_close)*> video_codec_ctx_cleanup(
video_codec_ctx.get(), avcodec_close);
+ // Used in decode_ycbcr_format().
+ is_mjpeg = video_codecpar->codec_id == AV_CODEC_ID_MJPEG;
+
// Open audio decoder, if we have audio.
AVCodecContextWithDeleter audio_codec_ctx;
if (audio_stream_index != -1) {
if (process_queued_commands(format_ctx.get(), pathname, last_modified, /*rewound=*/nullptr)) {
return true;
}
+ if (should_interrupt.load()) {
+ // Check as a failsafe, so that we don't need to rely on avio if we don't have to.
+ return false;
+ }
UniqueFrame audio_frame = audio_frame_allocator->alloc_frame();
AudioFormat audio_format;
int64_t audio_pts;
bool error;
AVFrameWithDeleter frame = decode_frame(format_ctx.get(), video_codec_ctx.get(), audio_codec_ctx.get(),
- pathname, video_stream_index, audio_stream_index, audio_frame.get(), &audio_format, &audio_pts, &error);
+ pathname, video_stream_index, audio_stream_index, subtitle_stream_index, audio_frame.get(), &audio_format, &audio_pts, &error);
if (error) {
return false;
}
if (frame == nullptr) {
// EOF. Loop back to the start if we can.
+ if (format_ctx->pb != nullptr && format_ctx->pb->seekable == 0) {
+ // Not seekable (but seemingly, sometimes av_seek_frame() would return 0 anyway,
+ // so don't try).
+ return true;
+ }
if (av_seek_frame(format_ctx.get(), /*stream_index=*/-1, /*timestamp=*/0, /*flags=*/0) < 0) {
fprintf(stderr, "%s: Rewind failed, not looping.\n", pathname.c_str());
return true;
if (last_pts == 0 && pts_origin == 0) {
pts_origin = frame->pts;
}
- next_frame_start = compute_frame_start(frame->pts, pts_origin, video_timebase, start, rate);
- if (first_frame && last_frame_was_connected) {
- // If reconnect took more than one second, this is probably a live feed,
- // and we should reset the resampler. (Or the rate is really, really low,
- // in which case a reset on the first frame is fine anyway.)
- if (duration<double>(next_frame_start - last_frame).count() >= 1.0) {
- last_frame_was_connected = false;
+ steady_clock::time_point now = steady_clock::now();
+ if (play_as_fast_as_possible) {
+ video_frame->received_timestamp = now;
+ audio_frame->received_timestamp = now;
+ next_frame_start = now;
+ } else {
+ next_frame_start = compute_frame_start(frame->pts, pts_origin, video_timebase, start, rate);
+ if (first_frame && last_frame_was_connected) {
+ // If reconnect took more than one second, this is probably a live feed,
+ // and we should reset the resampler. (Or the rate is really, really low,
+ // in which case a reset on the first frame is fine anyway.)
+ if (duration<double>(next_frame_start - last_frame).count() >= 1.0) {
+ last_frame_was_connected = false;
+ }
+ }
+ video_frame->received_timestamp = next_frame_start;
+
+ // The easiest way to get all the rate conversions etc. right is to move the
+ // audio PTS into the video PTS timebase and go from there. (We'll get some
+ // rounding issues, but they should not be a big problem.)
+ int64_t audio_pts_as_video_pts = av_rescale_q(audio_pts, audio_timebase, video_timebase);
+ audio_frame->received_timestamp = compute_frame_start(audio_pts_as_video_pts, pts_origin, video_timebase, start, rate);
+
+ if (audio_frame->len != 0) {
+ // The received timestamps in Nageru are measured after we've just received the frame.
+ // However, pts (especially audio pts) is at the _beginning_ of the frame.
+ // If we have locked audio, the distinction doesn't really matter, as pts is
+ // on a relative scale and a fixed offset is fine. But if we don't, we will have
+ // a different number of samples each time, which will cause huge audio jitter
+ // and throw off the resampler.
+ //
+ // In a sense, we should have compensated by adding the frame and audio lengths
+ // to video_frame->received_timestamp and audio_frame->received_timestamp respectively,
+ // but that would mean extra waiting in sleep_until(). All we need is that they
+ // are correct relative to each other, though (and to the other frames we send),
+ // so just align the end of the audio frame, and we're fine.
+ size_t num_samples = (audio_frame->len * 8) / audio_format.bits_per_sample / audio_format.num_channels;
+ double offset = double(num_samples) / OUTPUT_FREQUENCY -
+ double(video_format.frame_rate_den) / video_format.frame_rate_nom;
+ audio_frame->received_timestamp += duration_cast<steady_clock::duration>(duration<double>(offset));
}
- }
- video_frame->received_timestamp = next_frame_start;
-
- // The easiest way to get all the rate conversions etc. right is to move the
- // audio PTS into the video PTS timebase and go from there. (We'll get some
- // rounding issues, but they should not be a big problem.)
- int64_t audio_pts_as_video_pts = av_rescale_q(audio_pts, audio_timebase, video_timebase);
- audio_frame->received_timestamp = compute_frame_start(audio_pts_as_video_pts, pts_origin, video_timebase, start, rate);
-
- if (audio_frame->len != 0) {
- // The received timestamps in Nageru are measured after we've just received the frame.
- // However, pts (especially audio pts) is at the _beginning_ of the frame.
- // If we have locked audio, the distinction doesn't really matter, as pts is
- // on a relative scale and a fixed offset is fine. But if we don't, we will have
- // a different number of samples each time, which will cause huge audio jitter
- // and throw off the resampler.
- //
- // In a sense, we should have compensated by adding the frame and audio lengths
- // to video_frame->received_timestamp and audio_frame->received_timestamp respectively,
- // but that would mean extra waiting in sleep_until(). All we need is that they
- // are correct relative to each other, though (and to the other frames we send),
- // so just align the end of the audio frame, and we're fine.
- size_t num_samples = (audio_frame->len * 8) / audio_format.bits_per_sample / audio_format.num_channels;
- double offset = double(num_samples) / OUTPUT_FREQUENCY -
- double(video_format.frame_rate_den) / video_format.frame_rate_nom;
- audio_frame->received_timestamp += duration_cast<steady_clock::duration>(duration<double>(offset));
- }
- steady_clock::time_point now = steady_clock::now();
- if (duration<double>(now - next_frame_start).count() >= 0.1) {
- // If we don't have enough CPU to keep up, or if we have a live stream
- // where the initial origin was somehow wrong, we could be behind indefinitely.
- // In particular, this will give the audio resampler problems as it tries
- // to speed up to reduce the delay, hitting the low end of the buffer every time.
- fprintf(stderr, "%s: Playback %.0f ms behind, resetting time scale\n",
- pathname.c_str(),
- 1e3 * duration<double>(now - next_frame_start).count());
- pts_origin = frame->pts;
- start = next_frame_start = now;
- timecode += MAX_FPS * 2 + 1;
+ if (duration<double>(now - next_frame_start).count() >= 0.1) {
+ // If we don't have enough CPU to keep up, or if we have a live stream
+ // where the initial origin was somehow wrong, we could be behind indefinitely.
+ // In particular, this will give the audio resampler problems as it tries
+ // to speed up to reduce the delay, hitting the low end of the buffer every time.
+ fprintf(stderr, "%s: Playback %.0f ms behind, resetting time scale\n",
+ pathname.c_str(),
+ 1e3 * duration<double>(now - next_frame_start).count());
+ pts_origin = frame->pts;
+ start = next_frame_start = now;
+ timecode += MAX_FPS * 2 + 1;
+ }
+ }
+ bool finished_wakeup;
+ if (play_as_fast_as_possible) {
+ finished_wakeup = !producer_thread_should_quit.should_quit();
+ } else {
+ finished_wakeup = producer_thread_should_quit.sleep_until(next_frame_start);
}
- bool finished_wakeup = producer_thread_should_quit.sleep_until(next_frame_start);
if (finished_wakeup) {
if (audio_frame->len > 0) {
assert(audio_pts != -1);
// audio discontinuity.)
timecode += MAX_FPS * 2 + 1;
}
+ last_neutral_color = get_neutral_color(frame->metadata);
frame_callback(frame->pts, video_timebase, audio_pts, audio_timebase, timecode++,
video_frame.get_and_release(), 0, video_format,
audio_frame.get_and_release(), 0, audio_format);
start = compute_frame_start(last_pts, pts_origin, video_timebase, start, rate);
pts_origin = last_pts;
rate = cmd.new_rate;
+ play_as_fast_as_possible = (rate >= 10.0);
break;
}
}
} // namespace
AVFrameWithDeleter FFmpegCapture::decode_frame(AVFormatContext *format_ctx, AVCodecContext *video_codec_ctx, AVCodecContext *audio_codec_ctx,
- const std::string &pathname, int video_stream_index, int audio_stream_index,
+ const std::string &pathname, int video_stream_index, int audio_stream_index, int subtitle_stream_index,
FrameAllocator::Frame *audio_frame, AudioFormat *audio_format, int64_t *audio_pts, bool *error)
{
*error = false;
*error = true;
return AVFrameWithDeleter(nullptr);
}
+ } else if (pkt.stream_index == subtitle_stream_index) {
+ last_subtitle = string(reinterpret_cast<const char *>(pkt.data), pkt.size);
+ has_last_subtitle = true;
}
} else {
eof = true; // Or error, but ignore that for the time being.
audio_avframe->format != last_src_format ||
dst_format != last_dst_format ||
channel_layout != last_channel_layout ||
- av_frame_get_sample_rate(audio_avframe) != last_sample_rate) {
- avresample_free(&resampler);
- resampler = avresample_alloc_context();
+ audio_avframe->sample_rate != last_sample_rate) {
+ swr_free(&resampler);
+ resampler = swr_alloc_set_opts(nullptr,
+ /*out_ch_layout=*/AV_CH_LAYOUT_STEREO_DOWNMIX,
+ /*out_sample_fmt=*/dst_format,
+ /*out_sample_rate=*/OUTPUT_FREQUENCY,
+ /*in_ch_layout=*/channel_layout,
+ /*in_sample_fmt=*/AVSampleFormat(audio_avframe->format),
+ /*in_sample_rate=*/audio_avframe->sample_rate,
+ /*log_offset=*/0,
+ /*log_ctx=*/nullptr);
+
if (resampler == nullptr) {
fprintf(stderr, "Allocating resampler failed.\n");
- exit(1);
+ abort();
}
- av_opt_set_int(resampler, "in_channel_layout", channel_layout, 0);
- av_opt_set_int(resampler, "out_channel_layout", AV_CH_LAYOUT_STEREO_DOWNMIX, 0);
- av_opt_set_int(resampler, "in_sample_rate", av_frame_get_sample_rate(audio_avframe), 0);
- av_opt_set_int(resampler, "out_sample_rate", OUTPUT_FREQUENCY, 0);
- av_opt_set_int(resampler, "in_sample_fmt", audio_avframe->format, 0);
- av_opt_set_int(resampler, "out_sample_fmt", dst_format, 0);
-
- if (avresample_open(resampler) < 0) {
+ if (swr_init(resampler) < 0) {
fprintf(stderr, "Could not open resample context.\n");
- exit(1);
+ abort();
}
last_src_format = AVSampleFormat(audio_avframe->format);
last_dst_format = dst_format;
last_channel_layout = channel_layout;
- last_sample_rate = av_frame_get_sample_rate(audio_avframe);
+ last_sample_rate = audio_avframe->sample_rate;
}
size_t bytes_per_sample = (audio_format->bits_per_sample / 8) * 2;
size_t num_samples_room = (audio_frame->size - audio_frame->len) / bytes_per_sample;
uint8_t *data = audio_frame->data + audio_frame->len;
- int out_samples = avresample_convert(resampler, &data, 0, num_samples_room,
- const_cast<uint8_t **>(audio_avframe->data), audio_avframe->linesize[0], audio_avframe->nb_samples);
+ int out_samples = swr_convert(resampler, &data, num_samples_room,
+ const_cast<const uint8_t **>(audio_avframe->data), audio_avframe->nb_samples);
if (out_samples < 0) {
fprintf(stderr, "Audio conversion failed.\n");
- exit(1);
+ abort();
}
audio_frame->len += out_samples * bytes_per_sample;
video_format.stride = width;
}
video_format.frame_rate_nom = video_timebase.den;
- video_format.frame_rate_den = av_frame_get_pkt_duration(frame) * video_timebase.num;
+ video_format.frame_rate_den = frame->pkt_duration * video_timebase.num;
if (video_format.frame_rate_nom == 0 || video_format.frame_rate_den == 0) {
// Invalid frame rate.
video_format.frame_rate_nom = 60;
video_frame->len = (width * 2) * height;
const AVPixFmtDescriptor *desc = av_pix_fmt_desc_get(sws_dst_format);
- current_frame_ycbcr_format = decode_ycbcr_format(desc, frame);
+ current_frame_ycbcr_format = decode_ycbcr_format(desc, frame, is_mjpeg);
} else {
assert(pixel_format == bmusb::PixelFormat_8BitYCbCrPlanar);
const AVPixFmtDescriptor *desc = av_pix_fmt_desc_get(sws_dst_format);
video_frame->len = width * height + 2 * chroma_width * chroma_height;
- current_frame_ycbcr_format = decode_ycbcr_format(desc, frame);
+ current_frame_ycbcr_format = decode_ycbcr_format(desc, frame, is_mjpeg);
}
sws_scale(sws_ctx.get(), frame->data, frame->linesize, 0, frame->height, pic_data, linesizes);
projects / nageru / blobdiff