basic type mechanism working

1 files changed, 111 insertions, 48 deletions

diff --git a/ffmpeg/doc/examples/muxing.c b/ffmpeg/doc/examples/muxing.c
index 7305cc6..4cd3f65 100644
--- a/ffmpeg/doc/examples/muxing.c
+++ b/ffmpeg/doc/examples/muxing.c
@@ -34,9 +34,11 @@
 #include <string.h>
 #include <math.h>
 
+#include <libavutil/opt.h>
 #include <libavutil/mathematics.h>
 #include <libavformat/avformat.h>
 #include <libswscale/swscale.h>
+#include <libswresample/swresample.h>
 
 /* 5 seconds stream duration */
 #define STREAM_DURATION   200.0
@@ -46,13 +48,6 @@
 
 static int sws_flags = SWS_BICUBIC;
 
-/**************************************************************/
-/* audio output */
-
-static float t, tincr, tincr2;
-static int16_t *samples;
-static int audio_input_frame_size;
-
 /* Add an output stream. */
 static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec,
                             enum AVCodecID codec_id)
@@ -78,8 +73,7 @@ static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec,
 
     switch ((*codec)->type) {
     case AVMEDIA_TYPE_AUDIO:
-        st->id = 1;
-        c->sample_fmt  = AV_SAMPLE_FMT_S16;
+        c->sample_fmt  = AV_SAMPLE_FMT_FLTP;
         c->bit_rate    = 64000;
         c->sample_rate = 44100;
         c->channels    = 2;
@@ -127,8 +121,17 @@ static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec,
 /* audio output */
 
 static float t, tincr, tincr2;
-static int16_t *samples;
-static int audio_input_frame_size;
+
+static uint8_t **src_samples_data;
+static int       src_samples_linesize;
+static int       src_nb_samples;
+
+static int max_dst_nb_samples;
+uint8_t **dst_samples_data;
+int       dst_samples_linesize;
+int       dst_samples_size;
+
+struct SwrContext *swr_ctx = NULL;
 
 static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)
 {
@@ -150,17 +153,54 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)
     /* increment frequency by 110 Hz per second */
     tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;
 
-    if (c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)
-        audio_input_frame_size = 10000;
-    else
-        audio_input_frame_size = c->frame_size;
-    samples = av_malloc(audio_input_frame_size *
-                        av_get_bytes_per_sample(c->sample_fmt) *
-                        c->channels);
-    if (!samples) {
-        fprintf(stderr, "Could not allocate audio samples buffer\n");
+    src_nb_samples = c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE ?
+        10000 : c->frame_size;
+
+    ret = av_samples_alloc_array_and_samples(&src_samples_data, &src_samples_linesize, c->channels,
+                                             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();
+        if (!swr_ctx) {
+            fprintf(stderr, "Could not allocate resampler context\n");
+            exit(1);
+        }
+
+        /* set options */
+        av_opt_set_int       (swr_ctx, "in_channel_count",   c->channels,       0);
+        av_opt_set_int       (swr_ctx, "in_sample_rate",     c->sample_rate,    0);
+        av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt",      AV_SAMPLE_FMT_S16, 0);
+        av_opt_set_int       (swr_ctx, "out_channel_count",  c->channels,       0);
+        av_opt_set_int       (swr_ctx, "out_sample_rate",    c->sample_rate,    0);
+        av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt",     c->sample_fmt,     0);
+
+        /* initialize the resampling context */
+        if ((ret = swr_init(swr_ctx)) < 0) {
+            fprintf(stderr, "Failed to initialize the resampling context\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);
 }
 
 /* Prepare a 16 bit dummy audio frame of 'frame_size' samples and
@@ -184,19 +224,45 @@ static void write_audio_frame(AVFormatContext *oc, AVStream *st)
 {
     AVCodecContext *c;
     AVPacket pkt = { 0 }; // data and size must be 0;
-    AVFrame *frame = avcodec_alloc_frame();
-    int got_packet, ret;
+    AVFrame *frame = av_frame_alloc();
+    int got_packet, ret, dst_nb_samples;
 
     av_init_packet(&pkt);
     c = st->codec;
 
-    get_audio_frame(samples, audio_input_frame_size, c->channels);
-    frame->nb_samples = audio_input_frame_size;
+    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);
+        }
+    } else {
+        dst_nb_samples = src_nb_samples;
+    }
+
+    frame->nb_samples = dst_nb_samples;
     avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
-                             (uint8_t *)samples,
-                             audio_input_frame_size *
-                             av_get_bytes_per_sample(c->sample_fmt) *
-                             c->channels, 1);
+                             dst_samples_data[0], dst_samples_size, 0);
 
     ret = avcodec_encode_audio2(c, &pkt, frame, &got_packet);
     if (ret < 0) {
@@ -205,7 +271,7 @@ static void write_audio_frame(AVFormatContext *oc, AVStream *st)
     }
 
     if (!got_packet)
-        return;
+        goto freeframe;
 
     pkt.stream_index = st->index;
 
@@ -216,14 +282,19 @@ static void write_audio_frame(AVFormatContext *oc, AVStream *st)
                 av_err2str(ret));
         exit(1);
     }
-    avcodec_free_frame(&frame);
+freeframe:
+    av_frame_free(&frame);
 }
 
 static void close_audio(AVFormatContext *oc, AVStream *st)
 {
     avcodec_close(st->codec);
-
-    av_free(samples);
+    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(src_samples_data);
 }
 
 /**************************************************************/
@@ -246,7 +317,7 @@ static void open_video(AVFormatContext *oc, AVCodec *codec, AVStream *st)
     }
 
     /* allocate and init a re-usable frame */
-    frame = avcodec_alloc_frame();
+    frame = av_frame_alloc();
     if (!frame) {
         fprintf(stderr, "Could not allocate video frame\n");
         exit(1);
@@ -388,7 +459,7 @@ int main(int argc, char **argv)
     AVFormatContext *oc;
     AVStream *audio_st, *video_st;
     AVCodec *audio_codec, *video_codec;
-    double audio_pts, video_pts;
+    double audio_time, video_time;
     int ret;
 
     /* Initialize libavcodec, and register all codecs and formats. */
@@ -461,23 +532,15 @@ int main(int argc, char **argv)
         frame->pts = 0;
     for (;;) {
         /* Compute current audio and video time. */
-        if (audio_st)
-            audio_pts = (double)audio_st->pts.val * audio_st->time_base.num / audio_st->time_base.den;
-        else
-            audio_pts = 0.0;
-
-        if (video_st)
-            video_pts = (double)video_st->pts.val * video_st->time_base.num /
-                        video_st->time_base.den;
-        else
-            video_pts = 0.0;
-
-        if ((!audio_st || audio_pts >= STREAM_DURATION) &&
-            (!video_st || video_pts >= STREAM_DURATION))
+        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;
+
+        if ((!audio_st || audio_time >= STREAM_DURATION) &&
+            (!video_st || video_time >= STREAM_DURATION))
             break;
 
         /* write interleaved audio and video frames */
-        if (!video_st || (video_st && audio_st && audio_pts < video_pts)) {
+        if (!video_st || (video_st && audio_st && audio_time < video_time)) {
             write_audio_frame(oc, audio_st);
         } else {
             write_video_frame(oc, video_st);