doc/examples/demuxing_decoding: convert to new decoding API

doc/examples/demuxing_decoding.c

@@ -55,87 +55,93 @@ static AVPacket pkt;
 static int video_frame_count = 0;
 static int audio_frame_count = 0;
 
-static int decode_packet(int *got_frame, int cached)
+static int output_video_frame(AVFrame *frame)
+{
+    if (frame->width != width || frame->height != height ||
+        frame->format != pix_fmt) {
+        /* To handle this change, one could call av_image_alloc again and
+         * decode the following frames into another rawvideo file. */
+        fprintf(stderr, "Error: Width, height and pixel format have to be "
+                "constant in a rawvideo file, but the width, height or "
+                "pixel format of the input video changed:\n"
+                "old: width = %d, height = %d, format = %s\n"
+                "new: width = %d, height = %d, format = %s\n",
+                width, height, av_get_pix_fmt_name(pix_fmt),
+                frame->width, frame->height,
+                av_get_pix_fmt_name(frame->format));
+        return -1;
+    }
+
+    printf("video_frame n:%d coded_n:%d\n",
+           video_frame_count++, frame->coded_picture_number);
+
+    /* copy decoded frame to destination buffer:
+     * this is required since rawvideo expects non aligned data */
+    av_image_copy(video_dst_data, video_dst_linesize,
+                  (const uint8_t **)(frame->data), frame->linesize,
+                  pix_fmt, width, height);
+
+    /* write to rawvideo file */
+    fwrite(video_dst_data[0], 1, video_dst_bufsize, video_dst_file);
+    return 0;
+}
+
+static int output_audio_frame(AVFrame *frame)
+{
+    size_t unpadded_linesize = frame->nb_samples * av_get_bytes_per_sample(frame->format);
+    printf("audio_frame n:%d nb_samples:%d pts:%s\n",
+           audio_frame_count++, frame->nb_samples,
+           av_ts2timestr(frame->pts, &audio_dec_ctx->time_base));
+
+    /* Write the raw audio data samples of the first plane. This works
+     * fine for packed formats (e.g. AV_SAMPLE_FMT_S16). However,
+     * most audio decoders output planar audio, which uses a separate
+     * plane of audio samples for each channel (e.g. AV_SAMPLE_FMT_S16P).
+     * In other words, this code will write only the first audio channel
+     * in these cases.
+     * You should use libswresample or libavfilter to convert the frame
+     * to packed data. */
+    fwrite(frame->extended_data[0], 1, unpadded_linesize, audio_dst_file);
+
+    return 0;
+}
+
+static int decode_packet(AVCodecContext *dec, const AVPacket *pkt)
 {
     int ret = 0;
-    int decoded = pkt.size;
 
-    *got_frame = 0;
+    // submit the packet to the decoder
+    ret = avcodec_send_packet(dec, pkt);
+    if (ret < 0) {
+        fprintf(stderr, "Error submitting a packet for decoding (%s)\n", av_err2str(ret));
+        return ret;
+    }
 
-    if (pkt.stream_index == video_stream_idx) {
-        /* decode video frame */
-        ret = avcodec_decode_video2(video_dec_ctx, frame, got_frame, &pkt);
+    // get all the available frames from the decoder
+    while (ret >= 0) {
+        ret = avcodec_receive_frame(dec, frame);
         if (ret < 0) {
-            fprintf(stderr, "Error decoding video frame (%s)\n", av_err2str(ret));
-            return ret;
-        }
+            // those two return values are special and mean there is no output
+            // frame available, but there were no errors during decoding
+            if (ret == AVERROR_EOF || ret == AVERROR(EAGAIN))
+                return 0;
 
-        if (*got_frame) {
-
-            if (frame->width != width || frame->height != height ||
-                frame->format != pix_fmt) {
-                /* To handle this change, one could call av_image_alloc again and
-                 * decode the following frames into another rawvideo file. */
-                fprintf(stderr, "Error: Width, height and pixel format have to be "
-                        "constant in a rawvideo file, but the width, height or "
-                        "pixel format of the input video changed:\n"
-                        "old: width = %d, height = %d, format = %s\n"
-                        "new: width = %d, height = %d, format = %s\n",
-                        width, height, av_get_pix_fmt_name(pix_fmt),
-                        frame->width, frame->height,
-                        av_get_pix_fmt_name(frame->format));
-                return -1;
-            }
-
-            printf("video_frame%s n:%d coded_n:%d\n",
-                   cached ? "(cached)" : "",
-                   video_frame_count++, frame->coded_picture_number);
-
-            /* copy decoded frame to destination buffer:
-             * this is required since rawvideo expects non aligned data */
-            av_image_copy(video_dst_data, video_dst_linesize,
-                          (const uint8_t **)(frame->data), frame->linesize,
-                          pix_fmt, width, height);
-
-            /* write to rawvideo file */
-            fwrite(video_dst_data[0], 1, video_dst_bufsize, video_dst_file);
-        }
-    } else if (pkt.stream_index == audio_stream_idx) {
-        /* decode audio frame */
-        ret = avcodec_decode_audio4(audio_dec_ctx, frame, got_frame, &pkt);
-        if (ret < 0) {
-            fprintf(stderr, "Error decoding audio frame (%s)\n", av_err2str(ret));
+            fprintf(stderr, "Error during decoding (%s)\n", av_err2str(ret));
             return ret;
         }
-        /* Some audio decoders decode only part of the packet, and have to be
-         * called again with the remainder of the packet data.
-         * Sample: fate-suite/lossless-audio/luckynight-partial.shn
-         * Also, some decoders might over-read the packet. */
-        decoded = FFMIN(ret, pkt.size);
-
-        if (*got_frame) {
-            size_t unpadded_linesize = frame->nb_samples * av_get_bytes_per_sample(frame->format);
-            printf("audio_frame%s n:%d nb_samples:%d pts:%s\n",
-                   cached ? "(cached)" : "",
-                   audio_frame_count++, frame->nb_samples,
-                   av_ts2timestr(frame->pts, &audio_dec_ctx->time_base));
-
-            /* Write the raw audio data samples of the first plane. This works
-             * fine for packed formats (e.g. AV_SAMPLE_FMT_S16). However,
-             * most audio decoders output planar audio, which uses a separate
-             * plane of audio samples for each channel (e.g. AV_SAMPLE_FMT_S16P).
-             * In other words, this code will write only the first audio channel
-             * in these cases.
-             * You should use libswresample or libavfilter to convert the frame
-             * to packed data. */
-            fwrite(frame->extended_data[0], 1, unpadded_linesize, audio_dst_file);
-        }
-    }
 
-    if (*got_frame)
+        // write the frame data to output file
+        if (dec->codec->type == AVMEDIA_TYPE_VIDEO)
+            ret = output_video_frame(frame);
+        else
+            ret = output_audio_frame(frame);
+
         av_frame_unref(frame);
+        if (ret < 0)
+            return ret;
+    }
 
-    return decoded;
+    return 0;
 }
 
 static int open_codec_context(int *stream_idx,
@@ -221,7 +227,7 @@ static int get_format_from_sample_fmt(const char **fmt,
 
 int main (int argc, char **argv)
 {
-    int ret = 0, got_frame;
+    int ret = 0;
 
     if (argc != 4) {
         fprintf(stderr, "usage: %s  input_file video_output_file audio_output_file\n"
@@ -309,23 +315,22 @@ int main (int argc, char **argv)
 
     /* read frames from the file */
     while (av_read_frame(fmt_ctx, &pkt) >= 0) {
-        AVPacket orig_pkt = pkt;
-        do {
-            ret = decode_packet(&got_frame, 0);
-            if (ret < 0)
-                break;
-            pkt.data += ret;
-            pkt.size -= ret;
-        } while (pkt.size > 0);
-        av_packet_unref(&orig_pkt);
+        // check if the packet belongs to a stream we are interested in, otherwise
+        // skip it
+        if (pkt.stream_index == video_stream_idx)
+            ret = decode_packet(video_dec_ctx, &pkt);
+        else if (pkt.stream_index == audio_stream_idx)
+            ret = decode_packet(audio_dec_ctx, &pkt);
+        av_packet_unref(&pkt);
+        if (ret < 0)
+            break;
     }
 
-    /* flush cached frames */
-    pkt.data = NULL;
-    pkt.size = 0;
-    do {
-        decode_packet(&got_frame, 1);
-    } while (got_frame);
+    /* flush the decoders */
+    if (video_dec_ctx)
+        decode_packet(video_dec_ctx, NULL);
+    if (audio_dec_ctx)
+        decode_packet(audio_dec_ctx, NULL);
 
     printf("Demuxing succeeded.\n");