[FFmpeg-cvslog] avconv: use avcodec_decode_audio4() instead of avcodec_decode_audio3()

[Justin Ruggles]

ffmpeg | branch: master | Justin Ruggles <justin.ruggles at gmail.com> | Mon Nov 21 17:41:49 2011 -0500| [d1241ff3b289b49607910258e3e99a050a6df65a] | committer: Justin Ruggles

avconv: use avcodec_decode_audio4() instead of avcodec_decode_audio3()

> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=d1241ff3b289b49607910258e3e99a050a6df65a
---

 avconv.c |   55 ++++++++++++++++++++++++++-----------------------------
 1 files changed, 26 insertions(+), 29 deletions(-)

diff --git a/avconv.c b/avconv.c
index 8842b24..371a118 100644
--- a/avconv.c
+++ b/avconv.c
@@ -137,8 +137,6 @@ static uint8_t *audio_buf;
 static uint8_t *audio_out;
 static unsigned int allocated_audio_out_size, allocated_audio_buf_size;
 
-static void *samples;
-
 #define DEFAULT_PASS_LOGFILENAME_PREFIX "av2pass"
 
 typedef struct InputStream {
@@ -541,7 +539,6 @@ void exit_program(int ret)
     av_free(audio_buf);
     av_free(audio_out);
     allocated_audio_buf_size= allocated_audio_out_size= 0;
-    av_free(samples);
 
 #if CONFIG_AVFILTER
     avfilter_uninit();
@@ -737,14 +734,11 @@ static void generate_silence(uint8_t* buf, enum AVSampleFormat sample_fmt, size_
     memset(buf, fill_char, size);
 }
 
-static void do_audio_out(AVFormatContext *s,
-                         OutputStream *ost,
-                         InputStream *ist,
-                         unsigned char *buf, int size)
+static void do_audio_out(AVFormatContext *s, OutputStream *ost,
+                         InputStream *ist, AVFrame *decoded_frame)
 {
     uint8_t *buftmp;
     int64_t audio_out_size, audio_buf_size;
-    int64_t allocated_for_size= size;
 
     int size_out, frame_bytes, ret, resample_changed;
     AVCodecContext *enc= ost->st->codec;
@@ -752,6 +746,9 @@ static void do_audio_out(AVFormatContext *s,
     int osize = av_get_bytes_per_sample(enc->sample_fmt);
     int isize = av_get_bytes_per_sample(dec->sample_fmt);
     const int coded_bps = av_get_bits_per_sample(enc->codec->id);
+    uint8_t *buf = decoded_frame->data[0];
+    int size     = decoded_frame->nb_samples * dec->channels * isize;
+    int64_t allocated_for_size = size;
 
 need_realloc:
     audio_buf_size= (allocated_for_size + isize*dec->channels - 1) / (isize*dec->channels);
@@ -1620,39 +1617,40 @@ static void rate_emu_sleep(InputStream *ist)
 
 static int transcode_audio(InputStream *ist, AVPacket *pkt, int *got_output)
 {
-    static unsigned int samples_size = 0;
+    AVFrame *decoded_frame;
+    AVCodecContext *avctx = ist->st->codec;
     int bps = av_get_bytes_per_sample(ist->st->codec->sample_fmt);
-    uint8_t *decoded_data_buf  = NULL;
-    int      decoded_data_size = 0;
     int i, ret;
 
-    if (pkt && samples_size < FFMAX(pkt->size * bps, AVCODEC_MAX_AUDIO_FRAME_SIZE)) {
-        av_free(samples);
-        samples_size = FFMAX(pkt->size * bps, AVCODEC_MAX_AUDIO_FRAME_SIZE);
-        samples      = av_malloc(samples_size);
-    }
-    decoded_data_size = samples_size;
+    if (!(decoded_frame = avcodec_alloc_frame()))
+        return AVERROR(ENOMEM);
 
-    ret = avcodec_decode_audio3(ist->st->codec, samples, &decoded_data_size,
-                                pkt);
-    if (ret < 0)
+    ret = avcodec_decode_audio4(avctx, decoded_frame, got_output, pkt);
+    if (ret < 0) {
+        av_freep(&decoded_frame);
         return ret;
-    *got_output  = decoded_data_size > 0;
+    }
 
-    /* Some bug in mpeg audio decoder gives */
-    /* decoded_data_size < 0, it seems they are overflows */
     if (!*got_output) {
         /* no audio frame */
         return ret;
     }
 
-    decoded_data_buf = (uint8_t *)samples;
-    ist->next_pts   += ((int64_t)AV_TIME_BASE/bps * decoded_data_size) /
-                       (ist->st->codec->sample_rate * ist->st->codec->channels);
+    /* if the decoder provides a pts, use it instead of the last packet pts.
+       the decoder could be delaying output by a packet or more. */
+    if (decoded_frame->pts != AV_NOPTS_VALUE)
+        ist->next_pts = decoded_frame->pts;
+
+    /* increment next_pts to use for the case where the input stream does not
+       have timestamps or there are multiple frames in the packet */
+    ist->next_pts += ((int64_t)AV_TIME_BASE * decoded_frame->nb_samples) /
+                     avctx->sample_rate;
 
     // preprocess audio (volume)
     if (audio_volume != 256) {
-        switch (ist->st->codec->sample_fmt) {
+        int decoded_data_size = decoded_frame->nb_samples * avctx->channels * bps;
+        void *samples = decoded_frame->data[0];
+        switch (avctx->sample_fmt) {
         case AV_SAMPLE_FMT_U8:
         {
             uint8_t *volp = samples;
@@ -1713,8 +1711,7 @@ static int transcode_audio(InputStream *ist, AVPacket *pkt, int *got_output)
 
         if (!check_output_constraints(ist, ost) || !ost->encoding_needed)
             continue;
-        do_audio_out(output_files[ost->file_index].ctx, ost, ist,
-                     decoded_data_buf, decoded_data_size);
+        do_audio_out(output_files[ost->file_index].ctx, ost, ist, decoded_frame);
     }
     return ret;
 }