1 files changed, 136 insertions, 94 deletions

diff --git a/chromium/third_party/ffmpeg/doc/examples/muxing.c b/chromium/third_party/ffmpeg/doc/examples/muxing.c
index 4276114306e..ad8e027148d 100644
--- a/chromium/third_party/ffmpeg/doc/examples/muxing.c
+++ b/chromium/third_party/ffmpeg/doc/examples/muxing.c
@@ -24,9 +24,9 @@
  * @file
  * libavformat API example.
  *
- * Output a media file in any supported libavformat format.
- * The default codecs are used.
- * @example doc/examples/muxing.c
+ * Output a media file in any supported libavformat format. The default
+ * codecs are used.
+ * @example muxing.c
  */
 
 #include <stdlib.h>
@@ -36,18 +36,43 @@
 
 #include <libavutil/opt.h>
 #include <libavutil/mathematics.h>
+#include <libavutil/timestamp.h>
 #include <libavformat/avformat.h>
 #include <libswscale/swscale.h>
 #include <libswresample/swresample.h>
 
-/* 5 seconds stream duration */
-#define STREAM_DURATION   200.0
+static int audio_is_eof, video_is_eof;
+
+#define STREAM_DURATION   10.0
 #define STREAM_FRAME_RATE 25 /* 25 images/s */
-#define STREAM_NB_FRAMES  ((int)(STREAM_DURATION * STREAM_FRAME_RATE))
 #define STREAM_PIX_FMT    AV_PIX_FMT_YUV420P /* default pix_fmt */
 
 static int sws_flags = SWS_BICUBIC;
 
+static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt)
+{
+    AVRational *time_base = &fmt_ctx->streams[pkt->stream_index]->time_base;
+
+    printf("pts:%s pts_time:%s dts:%s dts_time:%s duration:%s duration_time:%s stream_index:%d\n",
+           av_ts2str(pkt->pts), av_ts2timestr(pkt->pts, time_base),
+           av_ts2str(pkt->dts), av_ts2timestr(pkt->dts, time_base),
+           av_ts2str(pkt->duration), av_ts2timestr(pkt->duration, time_base),
+           pkt->stream_index);
+}
+
+static int write_frame(AVFormatContext *fmt_ctx, const AVRational *time_base, AVStream *st, AVPacket *pkt)
+{
+    /* rescale output packet timestamp values from codec to stream timebase */
+    pkt->pts = av_rescale_q_rnd(pkt->pts, *time_base, st->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
+    pkt->dts = av_rescale_q_rnd(pkt->dts, *time_base, st->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
+    pkt->duration = av_rescale_q(pkt->duration, *time_base, st->time_base);
+    pkt->stream_index = st->index;
+
+    /* Write the compressed frame to the media file. */
+    log_packet(fmt_ctx, pkt);
+    return av_interleaved_write_frame(fmt_ctx, pkt);
+}
+
 /* Add an output stream. */
 static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec,
                             enum AVCodecID codec_id)
@@ -73,7 +98,8 @@ static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec,
 
     switch ((*codec)->type) {
     case AVMEDIA_TYPE_AUDIO:
-        c->sample_fmt  = AV_SAMPLE_FMT_FLTP;
+        c->sample_fmt  = (*codec)->sample_fmts ?
+            (*codec)->sample_fmts[0] : AV_SAMPLE_FMT_FLTP;
         c->bit_rate    = 64000;
         c->sample_rate = 44100;
         c->channels    = 2;
@@ -122,6 +148,7 @@ static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec,
 
 static float t, tincr, tincr2;
 
+AVFrame *audio_frame;
 static uint8_t **src_samples_data;
 static int       src_samples_linesize;
 static int       src_nb_samples;
@@ -130,6 +157,7 @@ static int max_dst_nb_samples;
 uint8_t **dst_samples_data;
 int       dst_samples_linesize;
 int       dst_samples_size;
+int samples_count;
 
 struct SwrContext *swr_ctx = NULL;
 
@@ -140,6 +168,13 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)
 
     c = st->codec;
 
+    /* allocate and init a re-usable frame */
+    audio_frame = av_frame_alloc();
+    if (!audio_frame) {
+        fprintf(stderr, "Could not allocate audio frame\n");
+        exit(1);
+    }
+
     /* open it */
     ret = avcodec_open2(c, codec, NULL);
     if (ret < 0) {
@@ -157,12 +192,17 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)
         10000 : c->frame_size;
 
     ret = av_samples_alloc_array_and_samples(&src_samples_data, &src_samples_linesize, c->channels,
-                                             src_nb_samples, c->sample_fmt, 0);
+                                             src_nb_samples, AV_SAMPLE_FMT_S16, 0);
     if (ret < 0) {
         fprintf(stderr, "Could not allocate source samples\n");
         exit(1);
     }
 
+    /* compute the number of converted samples: buffering is avoided
+     * ensuring that the output buffer will contain at least all the
+     * converted input samples */
+    max_dst_nb_samples = src_nb_samples;
+
     /* create resampler context */
     if (c->sample_fmt != AV_SAMPLE_FMT_S16) {
         swr_ctx = swr_alloc();
@@ -184,17 +224,15 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)
             fprintf(stderr, "Failed to initialize the resampling context\n");
             exit(1);
         }
-    }
 
-    /* compute the number of converted samples: buffering is avoided
-     * ensuring that the output buffer will contain at least all the
-     * converted input samples */
-    max_dst_nb_samples = src_nb_samples;
-    ret = av_samples_alloc_array_and_samples(&dst_samples_data, &dst_samples_linesize, c->channels,
-                                             max_dst_nb_samples, c->sample_fmt, 0);
-    if (ret < 0) {
-        fprintf(stderr, "Could not allocate destination samples\n");
-        exit(1);
+        ret = av_samples_alloc_array_and_samples(&dst_samples_data, &dst_samples_linesize, c->channels,
+                                                 max_dst_nb_samples, c->sample_fmt, 0);
+        if (ret < 0) {
+            fprintf(stderr, "Could not allocate destination samples\n");
+            exit(1);
+        }
+    } else {
+        dst_samples_data = src_samples_data;
     }
     dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, max_dst_nb_samples,
                                                   c->sample_fmt, 0);
@@ -217,77 +255,83 @@ static void get_audio_frame(int16_t *samples, int frame_size, int nb_channels)
     }
 }
 
-static void write_audio_frame(AVFormatContext *oc, AVStream *st)
+static void write_audio_frame(AVFormatContext *oc, AVStream *st, int flush)
 {
     AVCodecContext *c;
     AVPacket pkt = { 0 }; // data and size must be 0;
-    AVFrame *frame = av_frame_alloc();
     int got_packet, ret, dst_nb_samples;
 
     av_init_packet(&pkt);
     c = st->codec;
 
-    get_audio_frame((int16_t *)src_samples_data[0], src_nb_samples, c->channels);
-
-    /* convert samples from native format to destination codec format, using the resampler */
-    if (swr_ctx) {
-        /* compute destination number of samples */
-        dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, c->sample_rate) + src_nb_samples,
-                                        c->sample_rate, c->sample_rate, AV_ROUND_UP);
-        if (dst_nb_samples > max_dst_nb_samples) {
-            av_free(dst_samples_data[0]);
-            ret = av_samples_alloc(dst_samples_data, &dst_samples_linesize, c->channels,
-                                   dst_nb_samples, c->sample_fmt, 0);
-            if (ret < 0)
+    if (!flush) {
+        get_audio_frame((int16_t *)src_samples_data[0], src_nb_samples, c->channels);
+
+        /* convert samples from native format to destination codec format, using the resampler */
+        if (swr_ctx) {
+            /* compute destination number of samples */
+            dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, c->sample_rate) + src_nb_samples,
+                                            c->sample_rate, c->sample_rate, AV_ROUND_UP);
+            if (dst_nb_samples > max_dst_nb_samples) {
+                av_free(dst_samples_data[0]);
+                ret = av_samples_alloc(dst_samples_data, &dst_samples_linesize, c->channels,
+                                       dst_nb_samples, c->sample_fmt, 0);
+                if (ret < 0)
+                    exit(1);
+                max_dst_nb_samples = dst_nb_samples;
+                dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, dst_nb_samples,
+                                                              c->sample_fmt, 0);
+            }
+
+            /* convert to destination format */
+            ret = swr_convert(swr_ctx,
+                              dst_samples_data, dst_nb_samples,
+                              (const uint8_t **)src_samples_data, src_nb_samples);
+            if (ret < 0) {
+                fprintf(stderr, "Error while converting\n");
                 exit(1);
-            max_dst_nb_samples = dst_nb_samples;
-            dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, dst_nb_samples,
-                                                          c->sample_fmt, 0);
+            }
+        } else {
+            dst_nb_samples = src_nb_samples;
         }
 
-        /* convert to destination format */
-        ret = swr_convert(swr_ctx,
-                          dst_samples_data, dst_nb_samples,
-                          (const uint8_t **)src_samples_data, src_nb_samples);
-        if (ret < 0) {
-            fprintf(stderr, "Error while converting\n");
-            exit(1);
-        }
-    } else {
-        dst_samples_data[0] = src_samples_data[0];
-        dst_nb_samples = src_nb_samples;
+        audio_frame->nb_samples = dst_nb_samples;
+        audio_frame->pts = av_rescale_q(samples_count, (AVRational){1, c->sample_rate}, c->time_base);
+        avcodec_fill_audio_frame(audio_frame, c->channels, c->sample_fmt,
+                                 dst_samples_data[0], dst_samples_size, 0);
+        samples_count += dst_nb_samples;
     }
 
-    frame->nb_samples = dst_nb_samples;
-    avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
-                             dst_samples_data[0], dst_samples_size, 0);
-
-    ret = avcodec_encode_audio2(c, &pkt, frame, &got_packet);
+    ret = avcodec_encode_audio2(c, &pkt, flush ? NULL : audio_frame, &got_packet);
     if (ret < 0) {
         fprintf(stderr, "Error encoding audio frame: %s\n", av_err2str(ret));
         exit(1);
     }
 
-    if (!got_packet)
+    if (!got_packet) {
+        if (flush)
+            audio_is_eof = 1;
         return;
+    }
 
-    pkt.stream_index = st->index;
-
-    /* Write the compressed frame to the media file. */
-    ret = av_interleaved_write_frame(oc, &pkt);
-    if (ret != 0) {
+    ret = write_frame(oc, &c->time_base, st, &pkt);
+    if (ret < 0) {
         fprintf(stderr, "Error while writing audio frame: %s\n",
                 av_err2str(ret));
         exit(1);
     }
-    avcodec_free_frame(&frame);
 }
 
 static void close_audio(AVFormatContext *oc, AVStream *st)
 {
     avcodec_close(st->codec);
+    if (dst_samples_data != src_samples_data) {
+        av_free(dst_samples_data[0]);
+        av_free(dst_samples_data);
+    }
     av_free(src_samples_data[0]);
-    av_free(dst_samples_data[0]);
+    av_free(src_samples_data);
+    av_frame_free(&audio_frame);
 }
 
 /**************************************************************/
@@ -315,6 +359,9 @@ static void open_video(AVFormatContext *oc, AVCodec *codec, AVStream *st)
         fprintf(stderr, "Could not allocate video frame\n");
         exit(1);
     }
+    frame->format = c->pix_fmt;
+    frame->width = c->width;
+    frame->height = c->height;
 
     /* Allocate the encoded raw picture. */
     ret = avpicture_alloc(&dst_picture, c->pix_fmt, c->width, c->height);
@@ -361,17 +408,13 @@ static void fill_yuv_image(AVPicture *pict, int frame_index,
     }
 }
 
-static void write_video_frame(AVFormatContext *oc, AVStream *st)
+static void write_video_frame(AVFormatContext *oc, AVStream *st, int flush)
 {
     int ret;
     static struct SwsContext *sws_ctx;
     AVCodecContext *c = st->codec;
 
-    if (frame_count >= STREAM_NB_FRAMES) {
-        /* No more frames to compress. The codec has a latency of a few
-         * frames if using B-frames, so we get the last frames by
-         * passing the same picture again. */
-    } else {
+    if (!flush) {
         if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
             /* as we only generate a YUV420P picture, we must convert it
              * to the codec pixel format if needed */
@@ -394,7 +437,7 @@ static void write_video_frame(AVFormatContext *oc, AVStream *st)
         }
     }
 
-    if (oc->oformat->flags & AVFMT_RAWPICTURE) {
+    if (oc->oformat->flags & AVFMT_RAWPICTURE && !flush) {
         /* Raw video case - directly store the picture in the packet */
         AVPacket pkt;
         av_init_packet(&pkt);
@@ -411,23 +454,24 @@ static void write_video_frame(AVFormatContext *oc, AVStream *st)
         av_init_packet(&pkt);
 
         /* encode the image */
-        ret = avcodec_encode_video2(c, &pkt, frame, &got_packet);
+        frame->pts = frame_count;
+        ret = avcodec_encode_video2(c, &pkt, flush ? NULL : frame, &got_packet);
         if (ret < 0) {
             fprintf(stderr, "Error encoding video frame: %s\n", av_err2str(ret));
             exit(1);
         }
         /* If size is zero, it means the image was buffered. */
 
-        if (!ret && got_packet && pkt.size) {
-            pkt.stream_index = st->index;
-
-            /* Write the compressed frame to the media file. */
-            ret = av_interleaved_write_frame(oc, &pkt);
+        if (got_packet) {
+            ret = write_frame(oc, &c->time_base, st, &pkt);
         } else {
+            if (flush)
+                video_is_eof = 1;
             ret = 0;
         }
     }
-    if (ret != 0) {
+
+    if (ret < 0) {
         fprintf(stderr, "Error while writing video frame: %s\n", av_err2str(ret));
         exit(1);
     }
@@ -439,7 +483,7 @@ static void close_video(AVFormatContext *oc, AVStream *st)
     avcodec_close(st->codec);
     av_free(src_picture.data[0]);
     av_free(dst_picture.data[0]);
-    av_free(frame);
+    av_frame_free(&frame);
 }
 
 /**************************************************************/
@@ -453,7 +497,7 @@ int main(int argc, char **argv)
     AVStream *audio_st, *video_st;
     AVCodec *audio_codec, *video_codec;
     double audio_time, video_time;
-    int ret;
+    int flush, ret;
 
     /* Initialize libavcodec, and register all codecs and formats. */
     av_register_all();
@@ -477,9 +521,9 @@ int main(int argc, char **argv)
         printf("Could not deduce output format from file extension: using MPEG.\n");
         avformat_alloc_output_context2(&oc, NULL, "mpeg", filename);
     }
-    if (!oc) {
+    if (!oc)
         return 1;
-    }
+
     fmt = oc->oformat;
 
     /* Add the audio and video streams using the default format codecs
@@ -487,12 +531,10 @@ int main(int argc, char **argv)
     video_st = NULL;
     audio_st = NULL;
 
-    if (fmt->video_codec != AV_CODEC_ID_NONE) {
+    if (fmt->video_codec != AV_CODEC_ID_NONE)
         video_st = add_stream(oc, &video_codec, fmt->video_codec);
-    }
-    if (fmt->audio_codec != AV_CODEC_ID_NONE) {
+    if (fmt->audio_codec != AV_CODEC_ID_NONE)
         audio_st = add_stream(oc, &audio_codec, fmt->audio_codec);
-    }
 
     /* Now that all the parameters are set, we can open the audio and
      * video codecs and allocate the necessary encode buffers. */
@@ -521,23 +563,23 @@ int main(int argc, char **argv)
         return 1;
     }
 
-    if (frame)
-        frame->pts = 0;
-    for (;;) {
+    flush = 0;
+    while ((video_st && !video_is_eof) || (audio_st && !audio_is_eof)) {
         /* Compute current audio and video time. */
-        audio_time = audio_st ? audio_st->pts.val * av_q2d(audio_st->time_base) : 0.0;
-        video_time = video_st ? video_st->pts.val * av_q2d(video_st->time_base) : 0.0;
+        audio_time = (audio_st && !audio_is_eof) ? audio_st->pts.val * av_q2d(audio_st->time_base) : INFINITY;
+        video_time = (video_st && !video_is_eof) ? video_st->pts.val * av_q2d(video_st->time_base) : INFINITY;
 
-        if ((!audio_st || audio_time >= STREAM_DURATION) &&
-            (!video_st || video_time >= STREAM_DURATION))
-            break;
+        if (!flush &&
+            (!audio_st || audio_time >= STREAM_DURATION) &&
+            (!video_st || video_time >= STREAM_DURATION)) {
+            flush = 1;
+        }
 
         /* write interleaved audio and video frames */
-        if (!video_st || (video_st && audio_st && audio_time < video_time)) {
-            write_audio_frame(oc, audio_st);
-        } else {
-            write_video_frame(oc, video_st);
-            frame->pts += av_rescale_q(1, video_st->codec->time_base, video_st->time_base);
+        if (audio_st && !audio_is_eof && audio_time <= video_time) {
+            write_audio_frame(oc, audio_st, flush);
+        } else if (video_st && !video_is_eof && video_time < audio_time) {
+            write_video_frame(oc, video_st, flush);
         }
     }