codec_desc: Add missing DXV entry

[ffmpeg] / libavcodec / libmp3lame.c

diff --git a/libavcodec/libmp3lame.c b/libavcodec/libmp3lame.c
index ee76ff88128fc04362110b5b6217cc4ec3e170f5..966212dcae8146cbaf44303ba1df6f063122536b 100644 (file)
--- a/libavcodec/libmp3lame.c
+++ b/libavcodec/libmp3lame.c
@@ -48,6 +48,8 @@ typedef struct LAMEContext {
     int buffer_index;
     int buffer_size;
     int reservoir;
+    int joint_stereo;
+    int abr;
     float *samples_flt[2];
     AudioFrameQueue afq;
     AVFloatDSPContext fdsp;
@@ -59,7 +61,7 @@ static int realloc_buffer(LAMEContext *s)
     if (!s->buffer || s->buffer_size - s->buffer_index < BUFFER_SIZE) {
         int new_size = s->buffer_index + 2 * BUFFER_SIZE, err;
 
-        av_dlog(s->avctx, "resizing output buffer: %d -> %d\n", s->buffer_size,
+        ff_dlog(s->avctx, "resizing output buffer: %d -> %d\n", s->buffer_size,
                 new_size);
         if ((err = av_reallocp(&s->buffer, new_size)) < 0) {
             s->buffer_size = s->buffer_index = 0;
@@ -92,11 +94,11 @@ static av_cold int mp3lame_encode_init(AVCodecContext *avctx)
     s->avctx = avctx;
 
     /* initialize LAME and get defaults */
-    if ((s->gfp = lame_init()) == NULL)
+    if (!(s->gfp = lame_init()))
         return AVERROR(ENOMEM);
 
     lame_set_num_channels(s->gfp, avctx->channels);
-    lame_set_mode(s->gfp, avctx->channels > 1 ? JOINT_STEREO : MONO);
+    lame_set_mode(s->gfp, avctx->channels > 1 ? s->joint_stereo ? JOINT_STEREO : STEREO : MONO);
 
     /* sample rate */
     lame_set_in_samplerate (s->gfp, avctx->sample_rate);
@@ -109,12 +111,17 @@ static av_cold int mp3lame_encode_init(AVCodecContext *avctx)
         lame_set_quality(s->gfp, avctx->compression_level);
 
     /* rate control */
-    if (avctx->flags & CODEC_FLAG_QSCALE) {
+    if (avctx->flags & AV_CODEC_FLAG_QSCALE) { // VBR
         lame_set_VBR(s->gfp, vbr_default);
         lame_set_VBR_quality(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA);
     } else {
-        if (avctx->bit_rate)
-            lame_set_brate(s->gfp, avctx->bit_rate / 1000);
+        if (avctx->bit_rate) {
+            if (s->abr) {                   // ABR
+                lame_set_VBR(s->gfp, vbr_abr);
+                lame_set_VBR_mean_bitrate_kbps(s->gfp, avctx->bit_rate / 1000);
+            } else                          // CBR
+                lame_set_brate(s->gfp, avctx->bit_rate / 1000);
+        }
     }
 
     /* do not get a Xing VBR header frame from LAME */
@@ -130,7 +137,7 @@ static av_cold int mp3lame_encode_init(AVCodecContext *avctx)
     }
 
     /* get encoder delay */
-    avctx->delay = lame_get_encoder_delay(s->gfp) + 528 + 1;
+    avctx->initial_padding = lame_get_encoder_delay(s->gfp) + 528 + 1;
     ff_af_queue_init(avctx, &s->afq);
 
     avctx->frame_size  = lame_get_framesize(s->gfp);
@@ -152,7 +159,7 @@ static av_cold int mp3lame_encode_init(AVCodecContext *avctx)
     if (ret < 0)
         goto error;
 
-    avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
+    avpriv_float_dsp_init(&s->fdsp, avctx->flags & AV_CODEC_FLAG_BITEXACT);
 
     return 0;
 error:
@@ -175,6 +182,7 @@ static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
     MPADecodeHeader hdr;
     int len, ret, ch;
     int lame_result;
+    uint32_t h;
 
     if (frame) {
         switch (avctx->sample_fmt) {
@@ -230,12 +238,17 @@ static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
        determine the frame size. */
     if (s->buffer_index < 4)
         return 0;
-    if (avpriv_mpegaudio_decode_header(&hdr, AV_RB32(s->buffer))) {
+    h = AV_RB32(s->buffer);
+    if (ff_mpa_check_header(h) < 0) {
+        av_log(avctx, AV_LOG_ERROR, "Invalid mp3 header at start of buffer\n");
+        return AVERROR_BUG;
+    }
+    if (avpriv_mpegaudio_decode_header(&hdr, h)) {
         av_log(avctx, AV_LOG_ERROR, "free format output not supported\n");
         return -1;
     }
     len = hdr.frame_size;
-    av_dlog(avctx, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len,
+    ff_dlog(avctx, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len,
             s->buffer_index);
     if (len <= s->buffer_index) {
         if ((ret = ff_alloc_packet(avpkt, len))) {
@@ -260,6 +273,8 @@ static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
 #define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
 static const AVOption options[] = {
     { "reservoir", "Use bit reservoir.", OFFSET(reservoir), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 1, AE },
+    { "joint_stereo", "Use joint stereo.", OFFSET(joint_stereo), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 1, AE },
+    { "abr", "Use ABR", OFFSET(abr), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 1, AE },
     { NULL },
 };
 
@@ -288,7 +303,7 @@ AVCodec ff_libmp3lame_encoder = {
     .init                  = mp3lame_encode_init,
     .encode2               = mp3lame_encode_frame,
     .close                 = mp3lame_encode_close,
-    .capabilities          = CODEC_CAP_DELAY | CODEC_CAP_SMALL_LAST_FRAME,
+    .capabilities          = AV_CODEC_CAP_DELAY | AV_CODEC_CAP_SMALL_LAST_FRAME,
     .sample_fmts           = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P,
                                                              AV_SAMPLE_FMT_FLTP,
                                                              AV_SAMPLE_FMT_S16P,