Merge remote-tracking branch 'qatar/master'

* qatar/master: (36 commits)
  adpcmenc: Use correct frame_size for Yamaha ADPCM.
  avcodec: add ff_samples_to_time_base() convenience function to internal.h
  adx parser: set duration
  mlp parser: set duration instead of frame_size
  gsm parser: set duration
  mpegaudio parser: set duration instead of frame_size
  (e)ac3 parser: set duration instead of frame_size
  flac parser: set duration instead of frame_size
  avcodec: add duration field to AVCodecParserContext
  avutil: add av_rescale_q_rnd() to allow different rounding
  pnmdec: remove useless .pix_fmts
  libmp3lame: support float and s32 sample formats
  libmp3lame: renaming, rearrangement, alignment, and comments
  libmp3lame: use the LAME default bit rate
  libmp3lame: use avpriv_mpegaudio_decode_header() for output frame parsing
  libmp3lame: cosmetics: remove some pointless comments
  libmp3lame: convert some debugging code to av_dlog()
  libmp3lame: remove outdated comment.
  libmp3lame: do not set coded_frame->key_frame.
  libmp3lame: improve error handling in MP3lame_encode_init()
  ...

Conflicts:
	doc/APIchanges
	libavcodec/libmp3lame.c
	libavcodec/pcxenc.c
	libavcodec/pnmdec.c
	libavcodec/pnmenc.c
	libavcodec/sgienc.c
	libavcodec/utils.c
	libavformat/hls.c
	libavutil/avutil.h
	libswscale/x86/swscale_mmx.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>

1 files changed, 164 insertions, 189 deletions

diff --git a/libavcodec/libmp3lame.c b/libavcodec/libmp3lame.c
index d75183e9c0..e8accedc00 100644
--- a/libavcodec/libmp3lame.c
+++ b/libavcodec/libmp3lame.c
@@ -24,261 +24,227 @@
  * Interface to libmp3lame for mp3 encoding.
  */
 
+#include <lame/lame.h>
+
 #include "libavutil/intreadwrite.h"
 #include "libavutil/log.h"
 #include "libavutil/opt.h"
 #include "avcodec.h"
+#include "internal.h"
 #include "mpegaudio.h"
-#include <lame/lame.h>
+#include "mpegaudiodecheader.h"
 
 #define BUFFER_SIZE (7200 + 2 * MPA_FRAME_SIZE + MPA_FRAME_SIZE / 4+1000) // FIXME: Buffer size to small? Adding 1000 to make up for it.
-typedef struct Mp3AudioContext {
+
+typedef struct LAMEContext {
     AVClass *class;
+    AVCodecContext *avctx;
     lame_global_flags *gfp;
-    int stereo;
     uint8_t buffer[BUFFER_SIZE];
     int buffer_index;
-    struct {
-        int *left;
-        int *right;
-    } s32_data;
     int reservoir;
-} Mp3AudioContext;
+    void *planar_samples[2];
+} LAMEContext;
+
 
-static av_cold int MP3lame_encode_init(AVCodecContext *avctx)
+static av_cold int mp3lame_encode_close(AVCodecContext *avctx)
 {
-    Mp3AudioContext *s = avctx->priv_data;
+    LAMEContext *s = avctx->priv_data;
 
+    av_freep(&avctx->coded_frame);
+    av_freep(&s->planar_samples[0]);
+    av_freep(&s->planar_samples[1]);
+
+    lame_close(s->gfp);
+    return 0;
+}
+
+static av_cold int mp3lame_encode_init(AVCodecContext *avctx)
+{
+    LAMEContext *s = avctx->priv_data;
+    int ret;
+
+    s->avctx = avctx;
+
+    /* initialize LAME and get defaults */
+    if ((s->gfp = lame_init()) == NULL)
+        return AVERROR(ENOMEM);
+
+    /* channels */
     if (avctx->channels > 2) {
         av_log(avctx, AV_LOG_ERROR,
                "Invalid number of channels %d, must be <= 2\n", avctx->channels);
-        return AVERROR(EINVAL);
+        ret =  AVERROR(EINVAL);
+        goto error;
     }
+    lame_set_num_channels(s->gfp, avctx->channels);
+    lame_set_mode(s->gfp, avctx->channels > 1 ? JOINT_STEREO : MONO);
 
-    s->stereo = avctx->channels > 1 ? 1 : 0;
-
-    if ((s->gfp = lame_init()) == NULL)
-        goto err;
-    lame_set_in_samplerate(s->gfp, avctx->sample_rate);
+    /* sample rate */
+    lame_set_in_samplerate (s->gfp, avctx->sample_rate);
     lame_set_out_samplerate(s->gfp, avctx->sample_rate);
-    lame_set_num_channels(s->gfp, avctx->channels);
-    if (avctx->compression_level == FF_COMPRESSION_DEFAULT) {
+
+    /* algorithmic quality */
+    if (avctx->compression_level == FF_COMPRESSION_DEFAULT)
         lame_set_quality(s->gfp, 5);
-    } else {
+    else
         lame_set_quality(s->gfp, avctx->compression_level);
-    }
-    lame_set_mode(s->gfp, s->stereo ? JOINT_STEREO : MONO);
-    lame_set_brate(s->gfp, avctx->bit_rate / 1000);
+
+    /* rate control */
     if (avctx->flags & CODEC_FLAG_QSCALE) {
-        lame_set_brate(s->gfp, 0);
         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);
     }
-    lame_set_bWriteVbrTag(s->gfp,0);
-    lame_set_disable_reservoir(s->gfp, !s->reservoir);
-    if (lame_init_params(s->gfp) < 0)
-        goto err_close;
 
-    avctx->frame_size             = lame_get_framesize(s->gfp);
+    /* do not get a Xing VBR header frame from LAME */
+    lame_set_bWriteVbrTag(s->gfp,0);
 
-    if(!(avctx->coded_frame= avcodec_alloc_frame())) {
-        lame_close(s->gfp);
+    /* bit reservoir usage */
+    lame_set_disable_reservoir(s->gfp, !s->reservoir);
 
-        return AVERROR(ENOMEM);
+    /* set specified parameters */
+    if (lame_init_params(s->gfp) < 0) {
+        ret = -1;
+        goto error;
     }
 
-    if(AV_SAMPLE_FMT_S32 == avctx->sample_fmt && s->stereo) {
-        int nelem = 2 * avctx->frame_size;
-
-        if(! (s->s32_data.left = av_malloc(nelem * sizeof(int)))) {
-            av_freep(&avctx->coded_frame);
-            lame_close(s->gfp);
+    avctx->frame_size  = lame_get_framesize(s->gfp);
+    avctx->coded_frame = avcodec_alloc_frame();
+    if (!avctx->coded_frame) {
+        ret = AVERROR(ENOMEM);
+        goto error;
+    }
 
-            return AVERROR(ENOMEM);
+    /* sample format */
+    if (avctx->sample_fmt == AV_SAMPLE_FMT_S32 ||
+        avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {
+        int ch;
+        for (ch = 0; ch < avctx->channels; ch++) {
+            s->planar_samples[ch] = av_malloc(avctx->frame_size *
+                                              av_get_bytes_per_sample(avctx->sample_fmt));
+            if (!s->planar_samples[ch]) {
+                ret = AVERROR(ENOMEM);
+                goto error;
+            }
         }
-
-        s->s32_data.right = s->s32_data.left + avctx->frame_size;
     }
 
     return 0;
-
-err_close:
-    lame_close(s->gfp);
-err:
-    return -1;
+error:
+    mp3lame_encode_close(avctx);
+    return ret;
 }
 
-static const int sSampleRates[] = {
-    44100, 48000,  32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
-};
-
-static const int sBitRates[2][3][15] = {
-    {
-        { 0, 32, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416, 448 },
-        { 0, 32, 48, 56, 64,  80,  96,  112, 128, 160, 192, 224, 256, 320, 384 },
-        { 0, 32, 40, 48, 56,  64,  80,  96,  112, 128, 160, 192, 224, 256, 320 }
-    },
-    {
-        { 0, 32, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, 256 },
-        { 0,  8, 16, 24, 32, 40, 48,  56,  64,  80,  96, 112, 128, 144, 160 },
-        { 0,  8, 16, 24, 32, 40, 48,  56,  64,  80,  96, 112, 128, 144, 160 }
-    },
-};
-
-static const int sSamplesPerFrame[2][3] = {
-    { 384, 1152, 1152 },
-    { 384, 1152,  576 }
-};
-
-static const int sBitsPerSlot[3] = { 32, 8, 8 };
-
-static int mp3len(void *data, int *samplesPerFrame, int *sampleRate)
+#define DEINTERLEAVE(type, scale) do {                  \
+    int ch, i;                                          \
+    for (ch = 0; ch < s->avctx->channels; ch++) {       \
+        const type *input = samples;                    \
+        type      *output = s->planar_samples[ch];      \
+        input += ch;                                    \
+        for (i = 0; i < s->avctx->frame_size; i++) {    \
+            output[i] = *input * scale;                 \
+            input += s->avctx->channels;                \
+        }                                               \
+    }                                                   \
+} while (0)
+
+static int encode_frame_int16(LAMEContext *s, void *samples)
 {
-    uint32_t header  = AV_RB32(data);
-    int layerID      = 3 - ((header >> 17) & 0x03);
-    int bitRateID    = ((header >> 12) & 0x0f);
-    int sampleRateID = ((header >> 10) & 0x03);
-    int bitsPerSlot  = sBitsPerSlot[layerID];
-    int isPadded     = ((header >> 9) & 0x01);
-    static int const mode_tab[4] = { 2, 3, 1, 0 };
-    int mode    = mode_tab[(header >> 19) & 0x03];
-    int mpeg_id = mode > 0;
-    int temp0, temp1, bitRate;
-
-    if (((header >> 21) & 0x7ff) != 0x7ff || mode == 3 || layerID == 3 ||
-        sampleRateID == 3) {
-        return -1;
+    if (s->avctx->channels > 1) {
+        return lame_encode_buffer_interleaved(s->gfp, samples,
+                                              s->avctx->frame_size,
+                                              s->buffer + s->buffer_index,
+                                              BUFFER_SIZE - s->buffer_index);
+    } else {
+        return lame_encode_buffer(s->gfp, samples, NULL, s->avctx->frame_size,
+                                  s->buffer + s->buffer_index,
+                                  BUFFER_SIZE - s->buffer_index);
     }
+}
 
-    if (!samplesPerFrame)
-        samplesPerFrame = &temp0;
-    if (!sampleRate)
-        sampleRate      = &temp1;
+static int encode_frame_int32(LAMEContext *s, void *samples)
+{
+    DEINTERLEAVE(int32_t, 1);
 
-    //*isMono = ((header >>  6) & 0x03) == 0x03;
+    return lame_encode_buffer_int(s->gfp,
+                                  s->planar_samples[0], s->planar_samples[1],
+                                  s->avctx->frame_size,
+                                  s->buffer + s->buffer_index,
+                                  BUFFER_SIZE - s->buffer_index);
+}
 
-    *sampleRate      = sSampleRates[sampleRateID] >> mode;
-    bitRate          = sBitRates[mpeg_id][layerID][bitRateID] * 1000;
-    *samplesPerFrame = sSamplesPerFrame[mpeg_id][layerID];
-    //av_log(NULL, AV_LOG_DEBUG,
-    //       "sr:%d br:%d spf:%d l:%d m:%d\n",
-    //       *sampleRate, bitRate, *samplesPerFrame, layerID, mode);
+static int encode_frame_float(LAMEContext *s, void *samples)
+{
+    DEINTERLEAVE(float, 32768.0f);
 
-    return *samplesPerFrame * bitRate / (bitsPerSlot * *sampleRate) + isPadded;
+    return lame_encode_buffer_float(s->gfp,
+                                    s->planar_samples[0], s->planar_samples[1],
+                                    s->avctx->frame_size,
+                                    s->buffer + s->buffer_index,
+                                    BUFFER_SIZE - s->buffer_index);
 }
 
-static int MP3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame,
+static int mp3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame,
                                 int buf_size, void *data)
 {
-    Mp3AudioContext *s = avctx->priv_data;
+    LAMEContext *s = avctx->priv_data;
+    MPADecodeHeader hdr;
     int len;
     int lame_result;
 
-    /* lame 3.91 dies on '1-channel interleaved' data */
-
-    if (!data){
-        lame_result= lame_encode_flush(
-                s->gfp,
-                s->buffer + s->buffer_index,
-                BUFFER_SIZE - s->buffer_index
-                );
-#if 2147483647 == INT_MAX
-    }else if(AV_SAMPLE_FMT_S32 == avctx->sample_fmt){
-        if (s->stereo) {
-            int32_t *rp = data;
-            int32_t *mp = rp + 2*avctx->frame_size;
-            int *wpl = s->s32_data.left;
-            int *wpr = s->s32_data.right;
-
-            while (rp < mp) {
-                *wpl++ = *rp++;
-                *wpr++ = *rp++;
-            }
-
-            lame_result = lame_encode_buffer_int(
-                s->gfp,
-                s->s32_data.left,
-                s->s32_data.right,
-                avctx->frame_size,
-                s->buffer + s->buffer_index,
-                BUFFER_SIZE - s->buffer_index
-                );
-        } else {
-            lame_result = lame_encode_buffer_int(
-                s->gfp,
-                data,
-                data,
-                avctx->frame_size,
-                s->buffer + s->buffer_index,
-                BUFFER_SIZE - s->buffer_index
-                );
-        }
-#endif
-    }else{
-        if (s->stereo) {
-            lame_result = lame_encode_buffer_interleaved(
-                s->gfp,
-                data,
-                avctx->frame_size,
-                s->buffer + s->buffer_index,
-                BUFFER_SIZE - s->buffer_index
-                );
-        } else {
-            lame_result = lame_encode_buffer(
-                s->gfp,
-                data,
-                data,
-                avctx->frame_size,
-                s->buffer + s->buffer_index,
-                BUFFER_SIZE - s->buffer_index
-                );
+    if (data) {
+        switch (avctx->sample_fmt) {
+        case AV_SAMPLE_FMT_S16:
+            lame_result = encode_frame_int16(s, data);
+            break;
+        case AV_SAMPLE_FMT_S32:
+            lame_result = encode_frame_int32(s, data);
+            break;
+        case AV_SAMPLE_FMT_FLT:
+            lame_result = encode_frame_float(s, data);
+            break;
+        default:
+            return AVERROR_BUG;
         }
+    } else {
+        lame_result = lame_encode_flush(s->gfp, s->buffer + s->buffer_index,
+                                        BUFFER_SIZE - s->buffer_index);
     }
-
     if (lame_result < 0) {
         if (lame_result == -1) {
-            /* output buffer too small */
             av_log(avctx, AV_LOG_ERROR,
                    "lame: output buffer too small (buffer index: %d, free bytes: %d)\n",
                    s->buffer_index, BUFFER_SIZE - s->buffer_index);
         }
         return -1;
     }
-
     s->buffer_index += lame_result;
 
+    /* Move 1 frame from the LAME buffer to the output packet, if available.
+       We have to parse the first frame header in the output buffer to
+       determine the frame size. */
     if (s->buffer_index < 4)
         return 0;
-
-    len = mp3len(s->buffer, NULL, NULL);
-    //av_log(avctx, AV_LOG_DEBUG, "in:%d packet-len:%d index:%d\n",
-    //       avctx->frame_size, len, s->buffer_index);
+    if (avpriv_mpegaudio_decode_header(&hdr, AV_RB32(s->buffer))) {
+        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,
+            s->buffer_index);
     if (len <= s->buffer_index) {
         memcpy(frame, s->buffer, len);
         s->buffer_index -= len;
-
         memmove(s->buffer, s->buffer + len, s->buffer_index);
-        // FIXME fix the audio codec API, so we do not need the memcpy()
-        /*for(i=0; i<len; i++) {
-            av_log(avctx, AV_LOG_DEBUG, "%2X ", frame[i]);
-        }*/
         return len;
     } else
         return 0;
 }
 
-static av_cold int MP3lame_encode_close(AVCodecContext *avctx)
-{
-    Mp3AudioContext *s = avctx->priv_data;
-
-    av_freep(&s->s32_data.left);
-    av_freep(&avctx->coded_frame);
-
-    lame_close(s->gfp);
-    return 0;
-}
-
-#define OFFSET(x) offsetof(Mp3AudioContext, x)
+#define OFFSET(x) offsetof(LAMEContext, x)
 #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, { 1 }, 0, 1, AE },
@@ -292,21 +258,30 @@ static const AVClass libmp3lame_class = {
     .version    = LIBAVUTIL_VERSION_INT,
 };
 
+static const AVCodecDefault libmp3lame_defaults[] = {
+    { "b",          "0" },
+    { NULL },
+};
+
+static const int libmp3lame_sample_rates[] = {
+    44100, 48000,  32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
+};
+
 AVCodec ff_libmp3lame_encoder = {
     .name                  = "libmp3lame",
     .type                  = AVMEDIA_TYPE_AUDIO,
     .id                    = CODEC_ID_MP3,
-    .priv_data_size        = sizeof(Mp3AudioContext),
-    .init                  = MP3lame_encode_init,
-    .encode                = MP3lame_encode_frame,
-    .close                 = MP3lame_encode_close,
+    .priv_data_size        = sizeof(LAMEContext),
+    .init                  = mp3lame_encode_init,
+    .encode                = mp3lame_encode_frame,
+    .close                 = mp3lame_encode_close,
     .capabilities          = CODEC_CAP_DELAY,
-    .sample_fmts           = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16,
-#if 2147483647 == INT_MAX
-    AV_SAMPLE_FMT_S32,
-#endif
+    .sample_fmts           = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32,
+                                                             AV_SAMPLE_FMT_FLT,
+                                                             AV_SAMPLE_FMT_S16,
                                                              AV_SAMPLE_FMT_NONE },
-    .supported_samplerates = sSampleRates,
+    .supported_samplerates = libmp3lame_sample_rates,
     .long_name             = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
     .priv_class            = &libmp3lame_class,
+    .defaults              = libmp3lame_defaults,
 };