[libav-commits] avconv: use avcodec_encode_audio2()

Justin Ruggles git at libav.org
Mon Jan 16 03:43:52 CET 2012


Module: libav
Branch: master
Commit: ee458cb1fa3faa006785d725fb3e386174793000

Author:    Justin Ruggles <justin.ruggles at gmail.com>
Committer: Justin Ruggles <justin.ruggles at gmail.com>
Date:      Sat Dec 31 09:36:25 2011 -0500

avconv: use avcodec_encode_audio2()

---

 avconv.c |  205 ++++++++++++++++++++++++++++++++------------------------------
 1 files changed, 105 insertions(+), 100 deletions(-)

diff --git a/avconv.c b/avconv.c
index 817c0cf..2246851 100644
--- a/avconv.c
+++ b/avconv.c
@@ -141,8 +141,7 @@ static float dts_delta_threshold = 10;
 static int print_stats = 1;
 
 static uint8_t *audio_buf;
-static uint8_t *audio_out;
-static unsigned int allocated_audio_out_size, allocated_audio_buf_size;
+static unsigned int allocated_audio_buf_size;
 
 #define DEFAULT_PASS_LOGFILENAME_PREFIX "av2pass"
 
@@ -208,6 +207,7 @@ typedef struct OutputStream {
     AVBitStreamFilterContext *bitstream_filters;
     AVCodec *enc;
     int64_t max_frames;
+    AVFrame *output_frame;
 
     /* video only */
     int video_resample;
@@ -680,6 +680,13 @@ void exit_program(int ret)
         }
         output_streams[i].bitstream_filters = NULL;
 
+        if (output_streams[i].output_frame) {
+            AVFrame *frame = output_streams[i].output_frame;
+            if (frame->extended_data != frame->data)
+                av_freep(&frame->extended_data);
+            av_freep(&frame);
+        }
+
 #if CONFIG_AVFILTER
         av_freep(&output_streams[i].avfilter);
 #endif
@@ -705,8 +712,7 @@ void exit_program(int ret)
 
     uninit_opts();
     av_free(audio_buf);
-    av_free(audio_out);
-    allocated_audio_buf_size = allocated_audio_out_size = 0;
+    allocated_audio_buf_size = 0;
 
 #if CONFIG_AVFILTER
     avfilter_uninit();
@@ -907,18 +913,75 @@ static void generate_silence(uint8_t* buf, enum AVSampleFormat sample_fmt, size_
     memset(buf, fill_char, size);
 }
 
+static int encode_audio_frame(AVFormatContext *s, OutputStream *ost,
+                              const uint8_t *buf, int buf_size)
+{
+    AVCodecContext *enc = ost->st->codec;
+    AVFrame *frame = NULL;
+    AVPacket pkt;
+    int ret, got_packet;
+
+    av_init_packet(&pkt);
+    pkt.data = NULL;
+    pkt.size = 0;
+
+    if (buf) {
+        if (!ost->output_frame) {
+            ost->output_frame = avcodec_alloc_frame();
+            if (!ost->output_frame) {
+                av_log(NULL, AV_LOG_FATAL, "out-of-memory in encode_audio_frame()\n");
+                exit_program(1);
+            }
+        }
+        frame = ost->output_frame;
+        if (frame->extended_data != frame->data)
+            av_freep(&frame->extended_data);
+        avcodec_get_frame_defaults(frame);
+
+        frame->nb_samples  = buf_size /
+                             (enc->channels * av_get_bytes_per_sample(enc->sample_fmt));
+        if ((ret = avcodec_fill_audio_frame(frame, enc->channels, enc->sample_fmt,
+                                            buf, buf_size, 1)) < 0) {
+            av_log(NULL, AV_LOG_FATAL, "Audio encoding failed\n");
+            exit_program(1);
+        }
+    }
+
+    got_packet = 0;
+    if (avcodec_encode_audio2(enc, &pkt, frame, &got_packet) < 0) {
+        av_log(NULL, AV_LOG_FATAL, "Audio encoding failed\n");
+        exit_program(1);
+    }
+
+    if (got_packet) {
+        pkt.stream_index = ost->index;
+        if (pkt.pts != AV_NOPTS_VALUE)
+            pkt.pts      = av_rescale_q(pkt.pts,      enc->time_base, ost->st->time_base);
+        if (pkt.duration > 0)
+            pkt.duration = av_rescale_q(pkt.duration, enc->time_base, ost->st->time_base);
+
+        write_frame(s, &pkt, ost);
+
+        audio_size += pkt.size;
+    }
+
+    if (frame)
+        ost->sync_opts += frame->nb_samples;
+
+    return pkt.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 audio_buf_size;
 
-    int size_out, frame_bytes, ret, resample_changed;
+    int size_out, frame_bytes, resample_changed;
     AVCodecContext *enc = ost->st->codec;
     AVCodecContext *dec = ist->st->codec;
     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;
@@ -930,19 +993,13 @@ need_realloc:
     audio_buf_size  = FFMAX(audio_buf_size, enc->frame_size);
     audio_buf_size *= osize * enc->channels;
 
-    audio_out_size = FFMAX(audio_buf_size, enc->frame_size * osize * enc->channels);
-    if (coded_bps > 8 * osize)
-        audio_out_size = audio_out_size * coded_bps / (8*osize);
-    audio_out_size += FF_MIN_BUFFER_SIZE;
-
-    if (audio_out_size > INT_MAX || audio_buf_size > INT_MAX) {
+    if (audio_buf_size > INT_MAX) {
         av_log(NULL, AV_LOG_FATAL, "Buffer sizes too large\n");
         exit_program(1);
     }
 
     av_fast_malloc(&audio_buf, &allocated_audio_buf_size, audio_buf_size);
-    av_fast_malloc(&audio_out, &allocated_audio_out_size, audio_out_size);
-    if (!audio_buf || !audio_out) {
+    if (!audio_buf) {
         av_log(NULL, AV_LOG_FATAL, "Out of memory in do_audio_out\n");
         exit_program(1);
     }
@@ -1080,7 +1137,7 @@ need_realloc:
     }
 
     /* now encode as many frames as possible */
-    if (enc->frame_size > 1) {
+    if (!(enc->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)) {
         /* output resampled raw samples */
         if (av_fifo_realloc2(ost->fifo, av_fifo_size(ost->fifo) + size_out) < 0) {
             av_log(NULL, AV_LOG_FATAL, "av_fifo_realloc2() failed\n");
@@ -1091,62 +1148,11 @@ need_realloc:
         frame_bytes = enc->frame_size * osize * enc->channels;
 
         while (av_fifo_size(ost->fifo) >= frame_bytes) {
-            AVPacket pkt;
-            av_init_packet(&pkt);
-
             av_fifo_generic_read(ost->fifo, audio_buf, frame_bytes, NULL);
-
-            // FIXME pass ost->sync_opts as AVFrame.pts in avcodec_encode_audio()
-
-            ret = avcodec_encode_audio(enc, audio_out, audio_out_size,
-                                       (short *)audio_buf);
-            if (ret < 0) {
-                av_log(NULL, AV_LOG_FATAL, "Audio encoding failed\n");
-                exit_program(1);
-            }
-            audio_size += ret;
-            pkt.stream_index = ost->index;
-            pkt.data = audio_out;
-            pkt.size = ret;
-            if (enc->coded_frame && enc->coded_frame->pts != AV_NOPTS_VALUE)
-                pkt.pts = av_rescale_q(enc->coded_frame->pts, enc->time_base, ost->st->time_base);
-            pkt.flags |= AV_PKT_FLAG_KEY;
-            write_frame(s, &pkt, ost);
-
-            ost->sync_opts += enc->frame_size;
+            encode_audio_frame(s, ost, audio_buf, frame_bytes);
         }
     } else {
-        AVPacket pkt;
-        av_init_packet(&pkt);
-
-        ost->sync_opts += size_out / (osize * enc->channels);
-
-        /* output a pcm frame */
-        /* determine the size of the coded buffer */
-        size_out /= osize;
-        if (coded_bps)
-            size_out = size_out * coded_bps / 8;
-
-        if (size_out > audio_out_size) {
-            av_log(NULL, AV_LOG_FATAL, "Internal error, buffer size too small\n");
-            exit_program(1);
-        }
-
-        // FIXME pass ost->sync_opts as AVFrame.pts in avcodec_encode_audio()
-        ret = avcodec_encode_audio(enc, audio_out, size_out,
-                                   (short *)buftmp);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_FATAL, "Audio encoding failed\n");
-            exit_program(1);
-        }
-        audio_size += ret;
-        pkt.stream_index = ost->index;
-        pkt.data = audio_out;
-        pkt.size = ret;
-        if (enc->coded_frame && enc->coded_frame->pts != AV_NOPTS_VALUE)
-            pkt.pts = av_rescale_q(enc->coded_frame->pts, enc->time_base, ost->st->time_base);
-        pkt.flags |= AV_PKT_FLAG_KEY;
-        write_frame(s, &pkt, ost);
+        encode_audio_frame(s, ost, buftmp, size_out);
     }
 }
 
@@ -1620,6 +1626,7 @@ static void flush_encoders(OutputStream *ost_table, int nb_ostreams)
         OutputStream   *ost = &ost_table[i];
         AVCodecContext *enc = ost->st->codec;
         AVFormatContext *os = output_files[ost->file_index].ctx;
+        int stop_encoding = 0;
 
         if (!ost->encoding_needed)
             continue;
@@ -1633,41 +1640,35 @@ static void flush_encoders(OutputStream *ost_table, int nb_ostreams)
             AVPacket pkt;
             int fifo_bytes;
             av_init_packet(&pkt);
-            pkt.stream_index = ost->index;
+            pkt.data = NULL;
+            pkt.size = 0;
 
             switch (ost->st->codec->codec_type) {
             case AVMEDIA_TYPE_AUDIO:
                 fifo_bytes = av_fifo_size(ost->fifo);
-                ret = 0;
-                /* encode any samples remaining in fifo */
                 if (fifo_bytes > 0) {
-                    int osize = av_get_bytes_per_sample(enc->sample_fmt);
-                    int fs_tmp = enc->frame_size;
+                    /* encode any samples remaining in fifo */
+                    int frame_bytes = fifo_bytes;
 
                     av_fifo_generic_read(ost->fifo, audio_buf, fifo_bytes, NULL);
-                    if (enc->codec->capabilities & CODEC_CAP_SMALL_LAST_FRAME) {
-                        enc->frame_size = fifo_bytes / (osize * enc->channels);
-                    } else { /* pad */
-                        int frame_bytes = enc->frame_size*osize*enc->channels;
+
+                    /* pad last frame with silence if needed */
+                    if (!(enc->codec->capabilities & CODEC_CAP_SMALL_LAST_FRAME)) {
+                        frame_bytes = enc->frame_size * enc->channels *
+                                      av_get_bytes_per_sample(enc->sample_fmt);
                         if (allocated_audio_buf_size < frame_bytes)
                             exit_program(1);
                         generate_silence(audio_buf+fifo_bytes, enc->sample_fmt, frame_bytes - fifo_bytes);
                     }
-
-                    ret = avcodec_encode_audio(enc, bit_buffer, bit_buffer_size, (short *)audio_buf);
-                    pkt.duration = av_rescale((int64_t)enc->frame_size*ost->st->time_base.den,
-                                              ost->st->time_base.num, enc->sample_rate);
-                    enc->frame_size = fs_tmp;
-                }
-                if (ret <= 0) {
-                    ret = avcodec_encode_audio(enc, bit_buffer, bit_buffer_size, NULL);
-                }
-                if (ret < 0) {
-                    av_log(NULL, AV_LOG_FATAL, "Audio encoding failed\n");
-                    exit_program(1);
+                    encode_audio_frame(os, ost, audio_buf, frame_bytes);
+                } else {
+                    /* flush encoder with NULL frames until it is done
+                       returning packets */
+                    if (encode_audio_frame(os, ost, NULL, 0) == 0) {
+                        stop_encoding = 1;
+                        break;
+                    }
                 }
-                audio_size += ret;
-                pkt.flags  |= AV_PKT_FLAG_KEY;
                 break;
             case AVMEDIA_TYPE_VIDEO:
                 ret = avcodec_encode_video(enc, bit_buffer, bit_buffer_size, NULL);
@@ -1681,18 +1682,22 @@ static void flush_encoders(OutputStream *ost_table, int nb_ostreams)
                 if (ost->logfile && enc->stats_out) {
                     fprintf(ost->logfile, "%s", enc->stats_out);
                 }
+                if (ret <= 0) {
+                    stop_encoding = 1;
+                    break;
+                }
+                pkt.stream_index = ost->index;
+                pkt.data = bit_buffer;
+                pkt.size = ret;
+                if (enc->coded_frame && enc->coded_frame->pts != AV_NOPTS_VALUE)
+                    pkt.pts = av_rescale_q(enc->coded_frame->pts, enc->time_base, ost->st->time_base);
+                write_frame(os, &pkt, ost);
                 break;
             default:
-                ret = -1;
+                stop_encoding = 1;
             }
-
-            if (ret <= 0)
+            if (stop_encoding)
                 break;
-            pkt.data = bit_buffer;
-            pkt.size = ret;
-            if (enc->coded_frame && enc->coded_frame->pts != AV_NOPTS_VALUE)
-                pkt.pts = av_rescale_q(enc->coded_frame->pts, enc->time_base, ost->st->time_base);
-            write_frame(os, &pkt, ost);
         }
     }
 }



More information about the libav-commits mailing list