Merge remote-tracking branch 'qatar/master'

* qatar/master: (23 commits)
  aacenc: Fix identification padding when the bitstream is already aligned.
  aacenc: Write correct length for long identification strings.
  aud: remove unneeded field, audio_stream_index from context
  aud: fix time stamp calculation for ADPCM IMA WS
  aud: simplify header parsing
  aud: set pts_wrap_bits to 64.
  cosmetics: indentation
  aud: support Westwood SND1 audio in AUD files.
  adpcm_ima_ws: fix stereo decoding
  avcodec: add a new codec_id for CRYO APC IMA ADPCM.
  vqa: remove unused context fields, audio_samplerate and audio_bits
  vqa: clean up audio header parsing
  vqa: set time base to frame rate as coded in the header.
  vqa: set packet duration.
  vqa: use 1/sample_rate as the audio stream time base
  vqa: set stream start_time to 0.
  lavc: postpone the removal of AVCodecContext.request_channels.
  lavf: postpone removing av_close_input_file().
  lavc: postpone removing old audio encoding and decoding API
  avplay: remove the -er option.
  ...

Conflicts:
	Changelog
	libavcodec/version.h
	libavdevice/v4l.c

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

5 files changed, 101 insertions, 72 deletions

diff --git a/libavformat/apc.c b/libavformat/apc.c
index 5d7bb6f31c..1e942e69c9 100644
--- a/libavformat/apc.c
+++ b/libavformat/apc.c
@@ -44,7 +44,7 @@ static int apc_read_header(AVFormatContext *s, AVFormatParameters *ap)
         return AVERROR(ENOMEM);
 
     st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
-    st->codec->codec_id = CODEC_ID_ADPCM_IMA_WS;
+    st->codec->codec_id = CODEC_ID_ADPCM_IMA_APC;
 
     avio_rl32(pb); /* number of samples */
     st->codec->sample_rate = avio_rl32(pb);
diff --git a/libavformat/version.h b/libavformat/version.h
index aaa5212ff9..b4ee44f36c 100644
--- a/libavformat/version.h
+++ b/libavformat/version.h
@@ -123,7 +123,7 @@
 #define FF_API_SET_PTS_INFO            (LIBAVFORMAT_VERSION_MAJOR < 54)
 #endif
 #ifndef FF_API_CLOSE_INPUT_FILE
-#define FF_API_CLOSE_INPUT_FILE        (LIBAVFORMAT_VERSION_MAJOR < 54)
+#define FF_API_CLOSE_INPUT_FILE        (LIBAVFORMAT_VERSION_MAJOR < 55)
 #endif
 
 #endif /* AVFORMAT_VERSION_H */
diff --git a/libavformat/westwood_aud.c b/libavformat/westwood_aud.c
index b1eb768016..f083a86baf 100644
--- a/libavformat/westwood_aud.c
+++ b/libavformat/westwood_aud.c
@@ -41,15 +41,6 @@
 #define AUD_CHUNK_PREAMBLE_SIZE 8
 #define AUD_CHUNK_SIGNATURE 0x0000DEAF
 
-typedef struct WsAudDemuxContext {
-    int audio_samplerate;
-    int audio_channels;
-    int audio_bits;
-    enum CodecID audio_type;
-    int audio_stream_index;
-    int64_t audio_frame_counter;
-} WsAudDemuxContext;
-
 static int wsaud_probe(AVProbeData *p)
 {
     int field;
@@ -79,7 +70,7 @@ static int wsaud_probe(AVProbeData *p)
 
     /* note: only check for WS IMA (type 99) right now since there is no
      * support for type 1 */
-    if (p->buf[11] != 99)
+    if (p->buf[11] != 99 && p->buf[11] != 1)
         return 0;
 
     /* read ahead to the first audio chunk and validate the first header signature */
@@ -93,41 +84,44 @@ static int wsaud_probe(AVProbeData *p)
 static int wsaud_read_header(AVFormatContext *s,
                              AVFormatParameters *ap)
 {
-    WsAudDemuxContext *wsaud = s->priv_data;
     AVIOContext *pb = s->pb;
     AVStream *st;
     unsigned char header[AUD_HEADER_SIZE];
+    int sample_rate, channels, codec;
 
     if (avio_read(pb, header, AUD_HEADER_SIZE) != AUD_HEADER_SIZE)
         return AVERROR(EIO);
-    wsaud->audio_samplerate = AV_RL16(&header[0]);
-    if (header[11] == 99)
-        wsaud->audio_type = CODEC_ID_ADPCM_IMA_WS;
-    else
-        return AVERROR_INVALIDDATA;
 
-    /* flag 0 indicates stereo */
-    wsaud->audio_channels = (header[10] & 0x1) + 1;
-    /* flag 1 indicates 16 bit audio */
-    wsaud->audio_bits = (((header[10] & 0x2) >> 1) + 1) * 8;
+    sample_rate = AV_RL16(&header[0]);
+    channels    = (header[10] & 0x1) + 1;
+    codec       = header[11];
 
     /* initialize the audio decoder stream */
     st = avformat_new_stream(s, NULL);
     if (!st)
         return AVERROR(ENOMEM);
-    avpriv_set_pts_info(st, 33, 1, wsaud->audio_samplerate);
-    st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
-    st->codec->codec_id = wsaud->audio_type;
-    st->codec->codec_tag = 0;  /* no tag */
-    st->codec->channels = wsaud->audio_channels;
-    st->codec->sample_rate = wsaud->audio_samplerate;
-    st->codec->bits_per_coded_sample = wsaud->audio_bits;
-    st->codec->bit_rate = st->codec->channels * st->codec->sample_rate *
-        st->codec->bits_per_coded_sample / 4;
-    st->codec->block_align = st->codec->channels * st->codec->bits_per_coded_sample;
-
-    wsaud->audio_stream_index = st->index;
-    wsaud->audio_frame_counter = 0;
+
+    switch (codec) {
+    case  1:
+        if (channels != 1) {
+            av_log_ask_for_sample(s, "Stereo WS-SND1 is not supported.\n");
+            return AVERROR_PATCHWELCOME;
+        }
+        st->codec->codec_id = CODEC_ID_WESTWOOD_SND1;
+        break;
+    case 99:
+        st->codec->codec_id = CODEC_ID_ADPCM_IMA_WS;
+        st->codec->bits_per_coded_sample = 4;
+        st->codec->bit_rate = channels * sample_rate * 4;
+        break;
+    default:
+        av_log_ask_for_sample(s, "Unknown codec: %d\n", codec);
+        return AVERROR_PATCHWELCOME;
+    }
+    avpriv_set_pts_info(st, 64, 1, sample_rate);
+    st->codec->codec_type  = AVMEDIA_TYPE_AUDIO;
+    st->codec->channels    = channels;
+    st->codec->sample_rate = sample_rate;
 
     return 0;
 }
@@ -135,11 +129,11 @@ static int wsaud_read_header(AVFormatContext *s,
 static int wsaud_read_packet(AVFormatContext *s,
                              AVPacket *pkt)
 {
-    WsAudDemuxContext *wsaud = s->priv_data;
     AVIOContext *pb = s->pb;
     unsigned char preamble[AUD_CHUNK_PREAMBLE_SIZE];
     unsigned int chunk_size;
     int ret = 0;
+    AVStream *st = s->streams[0];
 
     if (avio_read(pb, preamble, AUD_CHUNK_PREAMBLE_SIZE) !=
         AUD_CHUNK_PREAMBLE_SIZE)
@@ -150,15 +144,30 @@ static int wsaud_read_packet(AVFormatContext *s,
         return AVERROR_INVALIDDATA;
 
     chunk_size = AV_RL16(&preamble[0]);
-    ret= av_get_packet(pb, pkt, chunk_size);
-    if (ret != chunk_size)
-        return AVERROR(EIO);
-    pkt->stream_index = wsaud->audio_stream_index;
-    pkt->pts = wsaud->audio_frame_counter;
-    pkt->pts /= wsaud->audio_samplerate;
 
-    /* 2 samples/byte, 1 or 2 samples per frame depending on stereo */
-    wsaud->audio_frame_counter += (chunk_size * 2) / wsaud->audio_channels;
+    if (st->codec->codec_id == CODEC_ID_WESTWOOD_SND1) {
+        /* For Westwood SND1 audio we need to add the output size and input
+           size to the start of the packet to match what is in VQA.
+           Specifically, this is needed to signal when a packet should be
+           decoding as raw 8-bit pcm or variable-size ADPCM. */
+        int out_size = AV_RL16(&preamble[2]);
+        if ((ret = av_new_packet(pkt, chunk_size + 4)))
+            return ret;
+        if ((ret = avio_read(pb, &pkt->data[4], chunk_size)) != chunk_size)
+            return ret < 0 ? ret : AVERROR(EIO);
+        AV_WL16(&pkt->data[0], out_size);
+        AV_WL16(&pkt->data[2], chunk_size);
+
+        pkt->duration = out_size;
+    } else {
+        ret = av_get_packet(pb, pkt, chunk_size);
+        if (ret != chunk_size)
+            return AVERROR(EIO);
+
+        /* 2 samples/byte, 1 or 2 samples per frame depending on stereo */
+        pkt->duration = (chunk_size * 2) / st->codec->channels;
+    }
+    pkt->stream_index = st->index;
 
     return ret;
 }
@@ -166,7 +175,6 @@ static int wsaud_read_packet(AVFormatContext *s,
 AVInputFormat ff_wsaud_demuxer = {
     .name           = "wsaud",
     .long_name      = NULL_IF_CONFIG_SMALL("Westwood Studios audio format"),
-    .priv_data_size = sizeof(WsAudDemuxContext),
     .read_probe     = wsaud_probe,
     .read_header    = wsaud_read_header,
     .read_packet    = wsaud_read_packet,
diff --git a/libavformat/westwood_vqa.c b/libavformat/westwood_vqa.c
index 4a851473b2..41dad8e8ab 100644
--- a/libavformat/westwood_vqa.c
+++ b/libavformat/westwood_vqa.c
@@ -51,18 +51,12 @@
 #define CMDS_TAG MKBETAG('C', 'M', 'D', 'S')
 
 #define VQA_HEADER_SIZE 0x2A
-#define VQA_FRAMERATE 15
 #define VQA_PREAMBLE_SIZE 8
 
 typedef struct WsVqaDemuxContext {
-    int audio_samplerate;
     int audio_channels;
-    int audio_bits;
-
     int audio_stream_index;
     int video_stream_index;
-
-    int64_t audio_frame_counter;
 } WsVqaDemuxContext;
 
 static int wsvqa_probe(AVProbeData *p)
@@ -89,12 +83,13 @@ static int wsvqa_read_header(AVFormatContext *s,
     unsigned char scratch[VQA_PREAMBLE_SIZE];
     unsigned int chunk_tag;
     unsigned int chunk_size;
+    int fps, version, flags, sample_rate, channels;
 
     /* initialize the video decoder stream */
     st = avformat_new_stream(s, NULL);
     if (!st)
         return AVERROR(ENOMEM);
-    avpriv_set_pts_info(st, 33, 1, VQA_FRAMERATE);
+    st->start_time = 0;
     wsvqa->video_stream_index = st->index;
     st->codec->codec_type = AVMEDIA_TYPE_VIDEO;
     st->codec->codec_id = CODEC_ID_WS_VQA;
@@ -114,34 +109,59 @@ static int wsvqa_read_header(AVFormatContext *s,
     }
     st->codec->width = AV_RL16(&header[6]);
     st->codec->height = AV_RL16(&header[8]);
+    fps = header[12];
+    if (fps < 1 || fps > 30) {
+        av_log(s, AV_LOG_ERROR, "invalid fps: %d\n", fps);
+        return AVERROR_INVALIDDATA;
+    }
+    avpriv_set_pts_info(st, 64, 1, fps);
 
     /* initialize the audio decoder stream for VQA v1 or nonzero samplerate */
-    if (AV_RL16(&header[24]) || (AV_RL16(&header[0]) == 1 && AV_RL16(&header[2]) == 1)) {
+    version     = AV_RL16(&header[ 0]);
+    flags       = AV_RL16(&header[ 2]);
+    sample_rate = AV_RL16(&header[24]);
+    channels    =          header[26];
+    if (sample_rate || (version == 1 && flags == 1)) {
         st = avformat_new_stream(s, NULL);
         if (!st)
             return AVERROR(ENOMEM);
-        avpriv_set_pts_info(st, 33, 1, VQA_FRAMERATE);
+        st->start_time = 0;
         st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
-        if (AV_RL16(&header[0]) == 1)
+
+        st->codec->extradata_size = VQA_HEADER_SIZE;
+        st->codec->extradata = av_mallocz(VQA_HEADER_SIZE + FF_INPUT_BUFFER_PADDING_SIZE);
+        if (!st->codec->extradata)
+            return AVERROR(ENOMEM);
+        memcpy(st->codec->extradata, header, VQA_HEADER_SIZE);
+
+        if (!sample_rate)
+            sample_rate = 22050;
+        st->codec->sample_rate = sample_rate;
+        avpriv_set_pts_info(st, 64, 1, st->codec->sample_rate);
+
+        if (!channels)
+            channels = 1;
+        st->codec->channels = channels;
+
+        switch (version) {
+        case 1:
             st->codec->codec_id = CODEC_ID_WESTWOOD_SND1;
-        else
+            break;
+        case 2:
+        case 3:
             st->codec->codec_id = CODEC_ID_ADPCM_IMA_WS;
-        st->codec->codec_tag = 0;  /* no tag */
-        st->codec->sample_rate = AV_RL16(&header[24]);
-        if (!st->codec->sample_rate)
-            st->codec->sample_rate = 22050;
-        st->codec->channels = header[26];
-        if (!st->codec->channels)
-            st->codec->channels = 1;
-        st->codec->bits_per_coded_sample = 16;
-        st->codec->bit_rate = st->codec->channels * st->codec->sample_rate *
-            st->codec->bits_per_coded_sample / 4;
-        st->codec->block_align = st->codec->channels * st->codec->bits_per_coded_sample;
+            st->codec->bits_per_coded_sample = 4;
+            st->codec->bit_rate = channels * sample_rate * 4;
+            break;
+        default:
+            /* NOTE: version 0 is supposedly raw pcm_u8 or pcm_s16le, but we do
+                     not have any samples to validate this */
+            av_log_ask_for_sample(s, "VQA version %d audio\n", version);
+            return AVERROR_PATCHWELCOME;
+        }
 
         wsvqa->audio_stream_index = st->index;
-        wsvqa->audio_samplerate = st->codec->sample_rate;
         wsvqa->audio_channels = st->codec->channels;
-        wsvqa->audio_frame_counter = 0;
     }
 
     /* there are 0 or more chunks before the FINF chunk; iterate until
@@ -208,13 +228,14 @@ static int wsvqa_read_packet(AVFormatContext *s,
             if (chunk_type == SND2_TAG) {
                 pkt->stream_index = wsvqa->audio_stream_index;
                 /* 2 samples/byte, 1 or 2 samples per frame depending on stereo */
-                wsvqa->audio_frame_counter += (chunk_size * 2) / wsvqa->audio_channels;
+                pkt->duration = (chunk_size * 2) / wsvqa->audio_channels;
             } else if(chunk_type == SND1_TAG) {
                 pkt->stream_index = wsvqa->audio_stream_index;
                 /* unpacked size is stored in header */
-                wsvqa->audio_frame_counter += AV_RL16(pkt->data) / wsvqa->audio_channels;
+                pkt->duration = AV_RL16(pkt->data) / wsvqa->audio_channels;
             } else {
                 pkt->stream_index = wsvqa->video_stream_index;
+                pkt->duration = 1;
             }
             /* stay on 16-bit alignment */
             if (skip_byte)
diff --git a/libavformat/yop.c b/libavformat/yop.c
index dac49d4dd2..eac3fb6707 100644
--- a/libavformat/yop.c
+++ b/libavformat/yop.c
@@ -72,7 +72,7 @@ static int yop_read_header(AVFormatContext *s, AVFormatParameters *ap)
     // Audio
     audio_dec               = audio_stream->codec;
     audio_dec->codec_type   = AVMEDIA_TYPE_AUDIO;
-    audio_dec->codec_id     = CODEC_ID_ADPCM_IMA_WS;
+    audio_dec->codec_id     = CODEC_ID_ADPCM_IMA_APC;
     audio_dec->channels     = 1;
     audio_dec->sample_rate  = 22050;