#include <X11/Xlib.h>
#include <assert.h>
#include <epoxy/egl.h>
-extern "C" {
-#include <libavcodec/avcodec.h>
-#include <libavformat/avformat.h>
-#include <libavresample/avresample.h>
-#include <libavutil/channel_layout.h>
-#include <libavutil/frame.h>
-#include <libavutil/rational.h>
-#include <libavutil/samplefmt.h>
-#include <libavutil/opt.h>
-}
#include <libdrm/drm_fourcc.h>
#include <stdio.h>
#include <stdlib.h>
#include <thread>
#include <utility>
+#include "audio_encoder.h"
#include "context.h"
#include "defs.h"
#include "flags.h"
class QuickSyncEncoderImpl {
public:
- QuickSyncEncoderImpl(QSurface *surface, const string &va_display, int width, int height, Mux *stream_mux);
+ QuickSyncEncoderImpl(const std::string &filename, QSurface *surface, const string &va_display, int width, int height, Mux *stream_mux, AudioEncoder *stream_audio_encoder, X264Encoder *x264_encoder);
~QuickSyncEncoderImpl();
void add_audio(int64_t pts, vector<float> audio);
bool begin_frame(GLuint *y_tex, GLuint *cbcr_tex);
RefCountedGLsync end_frame(int64_t pts, int64_t duration, const vector<RefCountedFrame> &input_frames);
void shutdown();
- void open_output_file(const std::string &filename);
- void close_output_file();
private:
struct storage_task {
return int64_t(ip_period - 1) * (TIMEBASE / MAX_FPS);
}
+ void open_output_file(const std::string &filename);
void encode_thread_func();
void encode_remaining_frames_as_p(int encoding_frame_num, int gop_start_display_frame_num, int64_t last_dts);
void add_packet_for_uncompressed_frame(int64_t pts, int64_t duration, const uint8_t *data);
void encode_frame(PendingFrame frame, int encoding_frame_num, int display_frame_num, int gop_start_display_frame_num,
int frame_type, int64_t pts, int64_t dts, int64_t duration);
void storage_task_thread();
- void encode_audio(const vector<float> &audio,
- vector<float> *audio_queue,
- int64_t audio_pts,
- AVCodecContext *ctx,
- AVAudioResampleContext *resampler,
- const vector<Mux *> &muxes);
- void encode_audio_one_frame(const float *audio,
- size_t num_samples, // In each channel.
- int64_t audio_pts,
- AVCodecContext *ctx,
- AVAudioResampleContext *resampler,
- const vector<Mux *> &muxes);
- void encode_last_audio(vector<float> *audio_queue,
- int64_t audio_pts,
- AVCodecContext *ctx,
- AVAudioResampleContext *resampler,
- const vector<Mux *> &muxes);
void encode_remaining_audio();
void storage_task_enqueue(storage_task task);
void save_codeddata(storage_task task);
map<int, PendingFrame> pending_video_frames; // under frame_queue_mutex
map<int64_t, vector<float>> pending_audio_frames; // under frame_queue_mutex
- int64_t last_audio_pts = 0; // The first pts after all audio we've encoded.
QSurface *surface;
- AVCodecContext *context_audio_file;
- AVCodecContext *context_audio_stream = nullptr; // nullptr = don't code separate audio for stream.
+ unique_ptr<AudioEncoder> file_audio_encoder;
+ AudioEncoder *stream_audio_encoder;
- AVAudioResampleContext *resampler_audio_file = nullptr;
- AVAudioResampleContext *resampler_audio_stream = nullptr;
-
- vector<float> audio_queue_file;
- vector<float> audio_queue_stream;
+ unique_ptr<FrameReorderer> reorderer;
+ X264Encoder *x264_encoder; // nullptr if not using x264.
Mux* stream_mux; // To HTTP.
unique_ptr<Mux> file_mux; // To local disk.
- AVFrame *audio_frame = nullptr;
- unique_ptr<FrameReorderer> reorderer;
- unique_ptr<X264Encoder> x264_encoder; // nullptr if not using x264.
-
Display *x11_display = nullptr;
// Encoder parameters
pending_audio_frames.erase(it);
}
- if (context_audio_stream) {
- encode_audio(audio, &audio_queue_file, audio_pts, context_audio_file, resampler_audio_file, { file_mux.get() });
- encode_audio(audio, &audio_queue_stream, audio_pts, context_audio_stream, resampler_audio_stream, { stream_mux });
- } else {
- encode_audio(audio, &audio_queue_file, audio_pts, context_audio_file, resampler_audio_file, { stream_mux, file_mux.get() });
- }
- last_audio_pts = audio_pts + audio.size() * TIMEBASE / (OUTPUT_FREQUENCY * 2);
+ file_audio_encoder->encode_audio(audio, audio_pts + global_delay());
+ stream_audio_encoder->encode_audio(audio, audio_pts + global_delay());
if (audio_pts == task.pts) break;
}
}
-void QuickSyncEncoderImpl::encode_audio(
- const vector<float> &audio,
- vector<float> *audio_queue,
- int64_t audio_pts,
- AVCodecContext *ctx,
- AVAudioResampleContext *resampler,
- const vector<Mux *> &muxes)
-{
- if (ctx->frame_size == 0) {
- // No queueing needed.
- assert(audio_queue->empty());
- assert(audio.size() % 2 == 0);
- encode_audio_one_frame(&audio[0], audio.size() / 2, audio_pts, ctx, resampler, muxes);
- return;
- }
-
- int64_t sample_offset = audio_queue->size();
-
- audio_queue->insert(audio_queue->end(), audio.begin(), audio.end());
- size_t sample_num;
- for (sample_num = 0;
- sample_num + ctx->frame_size * 2 <= audio_queue->size();
- sample_num += ctx->frame_size * 2) {
- int64_t adjusted_audio_pts = audio_pts + (int64_t(sample_num) - sample_offset) * TIMEBASE / (OUTPUT_FREQUENCY * 2);
- encode_audio_one_frame(&(*audio_queue)[sample_num],
- ctx->frame_size,
- adjusted_audio_pts,
- ctx,
- resampler,
- muxes);
- }
- audio_queue->erase(audio_queue->begin(), audio_queue->begin() + sample_num);
-}
-
-void QuickSyncEncoderImpl::encode_audio_one_frame(
- const float *audio,
- size_t num_samples,
- int64_t audio_pts,
- AVCodecContext *ctx,
- AVAudioResampleContext *resampler,
- const vector<Mux *> &muxes)
-{
- audio_frame->pts = audio_pts + global_delay();
- audio_frame->nb_samples = num_samples;
- audio_frame->channel_layout = AV_CH_LAYOUT_STEREO;
- audio_frame->format = ctx->sample_fmt;
- audio_frame->sample_rate = OUTPUT_FREQUENCY;
-
- if (av_samples_alloc(audio_frame->data, nullptr, 2, num_samples, ctx->sample_fmt, 0) < 0) {
- fprintf(stderr, "Could not allocate %ld samples.\n", num_samples);
- exit(1);
- }
-
- if (avresample_convert(resampler, audio_frame->data, 0, num_samples,
- (uint8_t **)&audio, 0, num_samples) < 0) {
- fprintf(stderr, "Audio conversion failed.\n");
- exit(1);
- }
-
- AVPacket pkt;
- av_init_packet(&pkt);
- pkt.data = nullptr;
- pkt.size = 0;
- int got_output = 0;
- avcodec_encode_audio2(ctx, &pkt, audio_frame, &got_output);
- if (got_output) {
- pkt.stream_index = 1;
- pkt.flags = 0;
- for (Mux *mux : muxes) {
- mux->add_packet(pkt, pkt.pts, pkt.dts);
- }
- }
-
- av_freep(&audio_frame->data[0]);
-
- av_frame_unref(audio_frame);
- av_free_packet(&pkt);
-}
-
-void QuickSyncEncoderImpl::encode_last_audio(
- vector<float> *audio_queue,
- int64_t audio_pts,
- AVCodecContext *ctx,
- AVAudioResampleContext *resampler,
- const vector<Mux *> &muxes)
-{
- if (!audio_queue->empty()) {
- // Last frame can be whatever size we want.
- assert(audio_queue->size() % 2 == 0);
- encode_audio_one_frame(&(*audio_queue)[0], audio_queue->size() / 2, audio_pts, ctx, resampler, muxes);
- audio_queue->clear();
- }
-
- if (ctx->codec->capabilities & AV_CODEC_CAP_DELAY) {
- // Collect any delayed frames.
- for ( ;; ) {
- int got_output = 0;
- AVPacket pkt;
- av_init_packet(&pkt);
- pkt.data = nullptr;
- pkt.size = 0;
- avcodec_encode_audio2(ctx, &pkt, nullptr, &got_output);
- if (!got_output) break;
-
- pkt.stream_index = 1;
- pkt.flags = 0;
- for (Mux *mux : muxes) {
- mux->add_packet(pkt, pkt.pts, pkt.dts);
- }
- av_free_packet(&pkt);
- }
- }
-}
// this is weird. but it seems to put a new frame onto the queue
void QuickSyncEncoderImpl::storage_task_enqueue(storage_task task)
namespace {
-void init_audio_encoder(const string &codec_name, int bit_rate, AVCodecContext **ctx, AVAudioResampleContext **resampler)
-{
- AVCodec *codec_audio = avcodec_find_encoder_by_name(codec_name.c_str());
- if (codec_audio == nullptr) {
- fprintf(stderr, "ERROR: Could not find codec '%s'\n", codec_name.c_str());
- exit(1);
- }
-
- AVCodecContext *context_audio = avcodec_alloc_context3(codec_audio);
- context_audio->bit_rate = bit_rate;
- context_audio->sample_rate = OUTPUT_FREQUENCY;
- context_audio->sample_fmt = codec_audio->sample_fmts[0];
- context_audio->channels = 2;
- context_audio->channel_layout = AV_CH_LAYOUT_STEREO;
- context_audio->time_base = AVRational{1, TIMEBASE};
- context_audio->flags |= CODEC_FLAG_GLOBAL_HEADER;
- if (avcodec_open2(context_audio, codec_audio, NULL) < 0) {
- fprintf(stderr, "Could not open codec '%s'\n", codec_name.c_str());
- exit(1);
- }
-
- *ctx = context_audio;
-
- *resampler = avresample_alloc_context();
- if (*resampler == nullptr) {
- fprintf(stderr, "Allocating resampler failed.\n");
- exit(1);
- }
-
- av_opt_set_int(*resampler, "in_channel_layout", AV_CH_LAYOUT_STEREO, 0);
- av_opt_set_int(*resampler, "out_channel_layout", AV_CH_LAYOUT_STEREO, 0);
- av_opt_set_int(*resampler, "in_sample_rate", OUTPUT_FREQUENCY, 0);
- av_opt_set_int(*resampler, "out_sample_rate", OUTPUT_FREQUENCY, 0);
- av_opt_set_int(*resampler, "in_sample_fmt", AV_SAMPLE_FMT_FLT, 0);
- av_opt_set_int(*resampler, "out_sample_fmt", context_audio->sample_fmt, 0);
-
- if (avresample_open(*resampler) < 0) {
- fprintf(stderr, "Could not open resample context.\n");
- exit(1);
- }
-}
-
} // namespace
-QuickSyncEncoderImpl::QuickSyncEncoderImpl(QSurface *surface, const string &va_display, int width, int height, Mux *stream_mux)
- : current_storage_frame(0), surface(surface), stream_mux(stream_mux), frame_width(width), frame_height(height)
+QuickSyncEncoderImpl::QuickSyncEncoderImpl(const std::string &filename, QSurface *surface, const string &va_display, int width, int height, Mux *stream_mux, AudioEncoder *stream_audio_encoder, X264Encoder *x264_encoder)
+ : current_storage_frame(0), surface(surface), stream_audio_encoder(stream_audio_encoder), x264_encoder(x264_encoder), stream_mux(stream_mux), frame_width(width), frame_height(height)
{
- init_audio_encoder(AUDIO_OUTPUT_CODEC_NAME, DEFAULT_AUDIO_OUTPUT_BIT_RATE, &context_audio_file, &resampler_audio_file);
-
- if (!global_flags.stream_audio_codec_name.empty()) {
- init_audio_encoder(global_flags.stream_audio_codec_name,
- global_flags.stream_audio_codec_bitrate, &context_audio_stream, &resampler_audio_stream);
- }
+ file_audio_encoder.reset(new AudioEncoder(AUDIO_OUTPUT_CODEC_NAME, DEFAULT_AUDIO_OUTPUT_BIT_RATE));
+ open_output_file(filename);
+ file_audio_encoder->add_mux(file_mux.get());
frame_width_mbaligned = (frame_width + 15) & (~15);
frame_height_mbaligned = (frame_height + 15) & (~15);
- audio_frame = av_frame_alloc();
-
//print_input();
if (global_flags.uncompressed_video_to_http ||
reorderer.reset(new FrameReorderer(ip_period - 1, frame_width, frame_height));
}
if (global_flags.x264_video_to_http) {
- x264_encoder.reset(new X264Encoder(stream_mux));
+ assert(x264_encoder != nullptr);
+ } else {
+ assert(x264_encoder == nullptr);
}
init_va(va_display);
QuickSyncEncoderImpl::~QuickSyncEncoderImpl()
{
shutdown();
- av_frame_free(&audio_frame);
- avresample_free(&resampler_audio_file);
- avresample_free(&resampler_audio_stream);
- avcodec_free_context(&context_audio_file);
- avcodec_free_context(&context_audio_stream);
}
bool QuickSyncEncoderImpl::begin_frame(GLuint *y_tex, GLuint *cbcr_tex)
frame_queue_nonempty.notify_all();
}
encode_thread.join();
- x264_encoder.reset();
{
unique_lock<mutex> lock(storage_task_queue_mutex);
storage_thread_should_quit = true;
release_encode();
deinit_va();
+ file_mux.reset();
is_shutdown = true;
}
exit(1);
}
- file_mux.reset(new Mux(avctx, frame_width, frame_height, Mux::CODEC_H264, context_audio_file->codec, TIMEBASE, DEFAULT_AUDIO_OUTPUT_BIT_RATE, nullptr));
-}
-
-void QuickSyncEncoderImpl::close_output_file()
-{
- file_mux.reset();
+ file_mux.reset(new Mux(avctx, frame_width, frame_height, Mux::CODEC_H264, file_audio_encoder->get_codec(), TIMEBASE, DEFAULT_AUDIO_OUTPUT_BIT_RATE, nullptr));
}
void QuickSyncEncoderImpl::encode_thread_func()
int64_t audio_pts = pending_frame.first;
vector<float> audio = move(pending_frame.second);
- if (context_audio_stream) {
- encode_audio(audio, &audio_queue_file, audio_pts, context_audio_file, resampler_audio_file, { file_mux.get() });
- encode_audio(audio, &audio_queue_stream, audio_pts, context_audio_stream, resampler_audio_stream, { stream_mux });
- } else {
- encode_audio(audio, &audio_queue_file, audio_pts, context_audio_file, resampler_audio_file, { stream_mux, file_mux.get() });
+ file_audio_encoder->encode_audio(audio, audio_pts + global_delay());
+ if (stream_audio_encoder) {
+ stream_audio_encoder->encode_audio(audio, audio_pts + global_delay());
}
- last_audio_pts = audio_pts + audio.size() * TIMEBASE / (OUTPUT_FREQUENCY * 2);
}
pending_audio_frames.clear();
// Encode any leftover audio in the queues, and also any delayed frames.
- if (context_audio_stream) {
- encode_last_audio(&audio_queue_file, last_audio_pts, context_audio_file, resampler_audio_file, { file_mux.get() });
- encode_last_audio(&audio_queue_stream, last_audio_pts, context_audio_stream, resampler_audio_stream, { stream_mux });
- } else {
- encode_last_audio(&audio_queue_file, last_audio_pts, context_audio_file, resampler_audio_file, { stream_mux, file_mux.get() });
- }
+ // Note: stream_audio_encoder is not owned by us, so don't call encode_last_audio().
+ file_audio_encoder->encode_last_audio();
}
void QuickSyncEncoderImpl::add_packet_for_uncompressed_frame(int64_t pts, int64_t duration, const uint8_t *data)
}
// Proxy object.
-QuickSyncEncoder::QuickSyncEncoder(QSurface *surface, const string &va_display, int width, int height, Mux *stream_mux)
- : impl(new QuickSyncEncoderImpl(surface, va_display, width, height, stream_mux)) {}
+QuickSyncEncoder::QuickSyncEncoder(const std::string &filename, QSurface *surface, const string &va_display, int width, int height, Mux *stream_mux, AudioEncoder *stream_audio_encoder, X264Encoder *x264_encoder)
+ : impl(new QuickSyncEncoderImpl(filename, surface, va_display, width, height, stream_mux, stream_audio_encoder, x264_encoder)) {}
// Must be defined here because unique_ptr<> destructor needs to know the impl.
QuickSyncEncoder::~QuickSyncEncoder() {}
{
impl->shutdown();
}
-
-void QuickSyncEncoder::open_output_file(const std::string &filename)
-{
- impl->open_output_file(filename);
-}
-
-void QuickSyncEncoder::close_output_file()
-{
- impl->close_output_file();
-}
projects / nageru / blobdiff