Add RTP

app/meson.build

@@ -23,6 +23,7 @@ src = [
     'src/options.c',
     'src/receiver.c',
     'src/recorder.c',
+    'src/rtp.c',
     'src/scrcpy.c',
     'src/screen.c',
     'src/server.c',

app/src/rtp.c

@@ -0,0 +1,319 @@
+#include "rtp.h"
+
+#include <assert.h>
+#include <libavcodec/avcodec.h>
+#include <libavformat/avformat.h>
+#include <libavutil/time.h>
+
+#include "util/log.h"
+
+/** Downcast packet_sink to rtp */
+#define DOWNCAST(SINK) container_of(SINK, struct sc_rtp, packet_sink)
+
+static const AVRational SCRCPY_TIME_BASE = {1, 1000000}; // timestamps in us
+
+static struct sc_rtp_packet *
+sc_rtp_packet_new(const AVPacket *packet) {
+    struct sc_rtp_packet *rtp = malloc(sizeof(*rtp));
+    if (!rtp) {
+        LOG_OOM();
+        return NULL;
+    }
+
+    rtp->packet = av_packet_alloc();
+    if (!rtp->packet) {
+        LOG_OOM();
+        free(rtp);
+        return NULL;
+    }
+
+    if (av_packet_ref(rtp->packet, packet)) {
+        av_packet_free(&rtp->packet);
+        free(rtp);
+        return NULL;
+    }
+
+    return rtp;
+}
+
+static void
+sc_rtp_packet_delete(struct sc_rtp_packet *rtp) {
+    av_packet_free(&rtp->packet);
+    free(rtp);
+}
+
+static void
+sc_rtp_queue_clear(struct sc_rtp_queue *queue) {
+    while (!sc_queue_is_empty(queue)) {
+        struct sc_rtp_packet *rtp;
+        sc_queue_take(queue, next, &rtp);
+        sc_rtp_packet_delete(rtp);
+    }
+}
+
+static bool
+sc_rtp_write_header(struct sc_rtp *rtp, const AVPacket *packet) {
+    AVStream *ostream = rtp->ctx->streams[0];
+
+    uint8_t *extradata = av_malloc(packet->size * sizeof(uint8_t));
+    if (!extradata) {
+        LOG_OOM();
+        return false;
+    }
+
+    // copy the first packet to the extra data
+    memcpy(extradata, packet->data, packet->size);
+
+    ostream->codecpar->extradata = extradata;
+    ostream->codecpar->extradata_size = packet->size;
+
+    int ret = avformat_write_header(rtp->ctx, NULL);
+    if (ret < 0) {
+        LOGE("Failed to write RTP header");
+        return false;
+    }
+
+    return true;
+}
+
+static void
+sc_rtp_rescale_packet(struct sc_rtp *rtp, AVPacket *packet) {
+    AVStream *ostream = rtp->ctx->streams[0];
+    av_packet_rescale_ts(packet, SCRCPY_TIME_BASE, ostream->time_base);
+}
+
+static bool
+sc_rtp_write(struct sc_rtp *rtp, AVPacket *packet) {
+    if (!rtp->header_written) {
+        if (packet->pts != AV_NOPTS_VALUE) {
+            LOGE("The first packet is not a config packet");
+            return false;
+        }
+        bool ok = sc_rtp_write_header(rtp, packet);
+        if (!ok) {
+            return false;
+        }
+        rtp->header_written = true;
+        return true;
+    }
+
+    if (packet->pts == AV_NOPTS_VALUE) {
+        // ignore config packets
+        return true;
+    }
+
+    sc_rtp_rescale_packet(rtp, packet);
+    return av_write_frame(rtp->ctx, packet) >= 0;
+}
+
+static int
+run_rtp(void *data) {
+    struct sc_rtp *rtp = data;
+
+    for (;;) {
+        sc_mutex_lock(&rtp->mutex);
+
+        while (!rtp->stopped && sc_queue_is_empty(&rtp->queue)) {
+            sc_cond_wait(&rtp->queue_cond, &rtp->mutex);
+        }
+
+        // if stopped is set, continue to process the remaining events (to
+        // finish the streaming) before actually stopping
+
+        if (rtp->stopped && sc_queue_is_empty(&rtp->queue)) {
+            sc_mutex_unlock(&rtp->mutex);
+            break;
+        }
+
+        struct sc_rtp_packet *pkt;
+        sc_queue_take(&rtp->queue, next, &pkt);
+
+        sc_mutex_unlock(&rtp->mutex);
+
+        bool ok = sc_rtp_write(rtp, pkt->packet);
+        sc_rtp_packet_delete(pkt);
+        if (!ok) {
+            LOGE("Could not send packet");
+
+            sc_mutex_lock(&rtp->mutex);
+            rtp->failed = true;
+            // discard pending packets
+            sc_rtp_queue_clear(&rtp->queue);
+            sc_mutex_unlock(&rtp->mutex);
+            break;
+        }
+    }
+
+    if (!rtp->failed) {
+        if (rtp->header_written) {
+            int ret = av_write_trailer(rtp->ctx);
+            if (ret < 0) {
+                LOGE("Failed to write RTP trailer");
+                rtp->failed = true;
+            }
+        } else {
+            // nothing has been sent
+            rtp->failed = true;
+        }
+    }
+
+    if (rtp->failed) {
+        LOGE("Streaming over RTP failed");
+    } else {
+        LOGI("Streaming over RTP complete");
+    }
+
+    LOGD("RTP streaming thread ended");
+
+    return 0;
+}
+
+static bool
+sc_rtp_open(struct sc_rtp *rtp, const AVCodec *input_codec) {
+    bool ok = sc_mutex_init(&rtp->mutex);
+    if (!ok) {
+        return false;
+    }
+
+    ok = sc_cond_init(&rtp->queue_cond);
+    if (!ok) {
+        goto error_mutex_destroy;
+    }
+
+    sc_queue_init(&rtp->queue);
+    rtp->stopped = false;
+    rtp->failed = false;
+    rtp->header_written = false;
+
+    int ret = avformat_alloc_output_context2(&rtp->ctx, NULL, "rtp",
+                                             rtp->out_url);
+    if (ret < 0) {
+        goto error_cond_destroy;
+    }
+
+    AVStream *ostream = avformat_new_stream(rtp->ctx, input_codec);
+    if (!ostream) {
+        goto error_avformat_free_context;
+    }
+
+    ostream->codecpar->codec_type = AVMEDIA_TYPE_VIDEO;
+    ostream->codecpar->codec_id = input_codec->id;
+    ostream->codecpar->width = rtp->declared_frame_size.width;
+    ostream->codecpar->height = rtp->declared_frame_size.height;
+
+    ret = avio_open(&rtp->ctx->pb, rtp->out_url, AVIO_FLAG_WRITE);
+    if (ret < 0) {
+        LOGE("Failed to open output: %s", rtp->out_url);
+        // ostream will be cleaned up during context cleaning
+        goto error_avformat_free_context;
+    }
+
+    LOGD("Starting RTP thread");
+    ok = sc_thread_create(&rtp->thread, run_rtp, "scrcpy-rtp", rtp);
+    if (!ok) {
+        LOGE("Could not start RTP thread");
+        goto error_avio_close;
+    }
+
+    LOGI("Streaming started to %s", rtp->out_url);
+
+    return true;
+
+error_avio_close:
+    avio_close(rtp->ctx->pb);
+error_avformat_free_context:
+    avformat_free_context(rtp->ctx);
+error_cond_destroy:
+    sc_cond_destroy(&rtp->queue_cond);
+error_mutex_destroy:
+    sc_mutex_destroy(&rtp->mutex);
+
+    return false;
+}
+
+static void
+sc_rtp_close(struct sc_rtp *rtp) {
+    sc_mutex_lock(&rtp->mutex);
+    rtp->stopped = true;
+    sc_cond_signal(&rtp->queue_cond);
+    sc_mutex_unlock(&rtp->mutex);
+
+    sc_thread_join(&rtp->thread, NULL);
+
+    avio_close(rtp->ctx->pb);
+    avformat_free_context(rtp->ctx);
+    sc_cond_destroy(&rtp->queue_cond);
+    sc_mutex_destroy(&rtp->mutex);
+}
+
+static bool
+sc_rtp_push(struct sc_rtp *rtp, const AVPacket *packet) {
+    sc_mutex_lock(&rtp->mutex);
+    assert(!rtp->stopped);
+
+    if (rtp->failed) {
+        // reject any new packet (this will stop the stream)
+        sc_mutex_unlock(&rtp->mutex);
+        return false;
+    }
+
+    struct sc_rtp_packet *pkt = sc_rtp_packet_new(packet);
+    if (!pkt) {
+        LOG_OOM();
+        sc_mutex_unlock(&rtp->mutex);
+        return false;
+    }
+
+    sc_queue_push(&rtp->queue, next, pkt);
+    sc_cond_signal(&rtp->queue_cond);
+
+    sc_mutex_unlock(&rtp->mutex);
+    return true;
+}
+
+static bool
+sc_rtp_packet_sink_open(struct sc_packet_sink *sink,
+                        const AVCodec *codec) {
+    struct sc_rtp *rtp = DOWNCAST(sink);
+    return sc_rtp_open(rtp, codec);
+}
+
+static void
+sc_rtp_packet_sink_close(struct sc_packet_sink *sink) {
+    struct sc_rtp *rtp = DOWNCAST(sink);
+    sc_rtp_close(rtp);
+}
+
+static bool
+sc_rtp_packet_sink_push(struct sc_packet_sink *sink,
+                        const AVPacket *packet) {
+    struct sc_rtp *rtp = DOWNCAST(sink);
+    return sc_rtp_push(rtp, packet);
+}
+
+bool
+sc_rtp_init(struct sc_rtp *rtp, const char *out_url,
+            struct sc_size declared_frame_size) {
+    rtp->out_url = strdup(out_url);
+    if (!rtp->out_url) {
+        LOG_OOM();
+        return false;
+    }
+
+    rtp->declared_frame_size = declared_frame_size;
+
+    static const struct sc_packet_sink_ops ops = {
+        .open = sc_rtp_packet_sink_open,
+        .close = sc_rtp_packet_sink_close,
+        .push = sc_rtp_packet_sink_push,
+    };
+
+    rtp->packet_sink.ops = &ops;
+
+    return true;
+}
+
+void
+sc_rtp_destroy(struct sc_rtp *rtp) {
+    free(rtp->out_url);
+}

app/src/rtp.h

@@ -0,0 +1,44 @@
+#ifndef SC_RTP_H
+#define SC_RTP_H
+
+#include "common.h"
+
+#include <stdbool.h>
+#include <libavformat/avformat.h>
+
+#include "coords.h"
+#include "trait/packet_sink.h"
+#include "util/queue.h"
+#include "util/thread.h"
+
+struct sc_rtp_packet {
+    AVPacket *packet;
+    struct sc_rtp_packet *next;
+};
+
+struct sc_rtp_queue SC_QUEUE(struct sc_rtp_packet);
+
+struct sc_rtp {
+    struct sc_packet_sink packet_sink; // packet sink trait;
+
+    char *out_url;
+    AVFormatContext *ctx;
+    struct sc_size declared_frame_size;
+    bool header_written;
+
+    sc_thread thread;
+    sc_mutex mutex;
+    sc_cond queue_cond;
+    bool stopped; // set on rtp_close()
+    bool failed; // set on packet write failure
+    struct sc_rtp_queue queue;
+};
+
+bool
+sc_rtp_init(struct sc_rtp *rtp, const char *out_url,
+            struct sc_size declared_frame_size);
+
+void
+sc_rtp_destroy(struct sc_rtp *rtp);
+
+#endif