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https://git.ffmpeg.org/ffmpeg.git
synced 2024-12-25 08:42:39 +00:00
- Fixed AC3 decoding for 5:1 AC3 streams. Now when calling av_audio_decode for
AC3 set avcodec_context->channels to the desired number channels, if the setting is 0 AC3 decoder will set it to the channels found in the stream. - Changed ffmpeg to cope with the new "way" of AC3 decoding. - ASF muxer now uses Tickers for PTS calculations. Originally committed as revision 393 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Juanjo
parent
9f862d1133
commit
e0d2714adc
15
ffmpeg.c
15
ffmpeg.c
@ -740,12 +740,21 @@ static int av_encode(AVFormatContext **output_files,
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if (codec->channels == icodec->channels &&
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codec->sample_rate == icodec->sample_rate) {
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ost->audio_resample = 0;
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} else {
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if (codec->channels != icodec->channels &&
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icodec->codec_id == CODEC_ID_AC3) {
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/* Special case for 5:1 AC3 input */
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/* and mono or stereo output */
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ost->audio_resample = 0;
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/* Request specific number of channels */
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icodec->channels = codec->channels;
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} else {
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ost->audio_resample = 1;
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ost->resample = audio_resample_init(codec->channels, icodec->channels,
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codec->sample_rate,
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icodec->sample_rate);
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}
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}
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ist->decoding_needed = 1;
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ost->encoding_needed = 1;
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}
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@ -1626,6 +1635,7 @@ void opt_input_file(const char *filename)
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AVCodecContext *enc = &ic->streams[i]->codec;
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switch(enc->codec_type) {
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case CODEC_TYPE_AUDIO:
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//fprintf(stderr, "\nInput Audio channels: %d", enc->channels);
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audio_channels = enc->channels;
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audio_sample_rate = enc->sample_rate;
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break;
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@ -1789,6 +1799,11 @@ void opt_output_file(const char *filename)
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audio_enc->bit_rate = audio_bit_rate;
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audio_enc->sample_rate = audio_sample_rate;
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/* For audio codecs other than AC3 we limit */
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/* the number of coded channels to stereo */
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if (audio_channels > 2 && codec_id != CODEC_ID_AC3) {
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audio_enc->channels = 2;
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} else
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audio_enc->channels = audio_channels;
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oc->streams[nb_streams] = st;
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nb_streams++;
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22
libav/asf.c
22
libav/asf.c
@ -18,6 +18,7 @@
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*/
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#include "avformat.h"
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#include "avi.h"
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#include "tick.h"
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#define PACKET_SIZE 3200
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#define PACKET_HEADER_SIZE 12
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@ -26,6 +27,7 @@
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typedef struct {
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int num;
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int seq;
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Ticker pts_ticker;
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/* use for reading */
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AVPacket pkt;
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int frag_offset;
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@ -283,6 +285,7 @@ static int asf_write_header1(AVFormatContext *s, INT64 file_size, INT64 data_chu
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/* stream headers */
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for(n=0;n<s->nb_streams;n++) {
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ASFStream *stream = &asf->streams[n];
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enc = &s->streams[n]->codec;
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asf->streams[n].num = n + 1;
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asf->streams[n].seq = 0;
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@ -292,12 +295,20 @@ static int asf_write_header1(AVFormatContext *s, INT64 file_size, INT64 data_chu
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wav_extra_size = 0;
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extra_size = 18 + wav_extra_size;
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extra_size2 = 0;
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/* Init the ticker */
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ticker_init(&stream->pts_ticker,
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enc->sample_rate,
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1000 * enc->frame_size);
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break;
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default:
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case CODEC_TYPE_VIDEO:
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wav_extra_size = 0;
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extra_size = 0x33;
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extra_size2 = 0;
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/* Init the ticker */
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ticker_init(&stream->pts_ticker,
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enc->frame_rate,
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1000 * FRAME_RATE_BASE);
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break;
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}
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@ -543,26 +554,27 @@ static int asf_write_packet(AVFormatContext *s, int stream_index,
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UINT8 *buf, int size, int force_pts)
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{
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ASFContext *asf = s->priv_data;
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ASFStream *stream;
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int timestamp;
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INT64 duration;
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AVCodecContext *codec;
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codec = &s->streams[stream_index]->codec;
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stream = &asf->streams[stream_index];
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if (codec->codec_type == CODEC_TYPE_AUDIO) {
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timestamp = (int)((float)codec->frame_number * codec->frame_size * 1000.0 /
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codec->sample_rate);
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timestamp = (int)ticker_tick(&stream->pts_ticker, codec->frame_number);
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duration = (codec->frame_number * codec->frame_size * INT64_C(10000000)) /
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codec->sample_rate;
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} else {
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timestamp = (int)((float)codec->frame_number * 1000.0 * FRAME_RATE_BASE /
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codec->frame_rate);
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timestamp = (int)ticker_tick(&stream->pts_ticker, codec->frame_number);
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duration = codec->frame_number *
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((INT64_C(10000000) * FRAME_RATE_BASE) / codec->frame_rate);
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}
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if (duration > asf->duration)
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asf->duration = duration;
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put_frame(s, &asf->streams[stream_index], (int)timestamp, buf, size);
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put_frame(s, stream, timestamp, buf, size);
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return 0;
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}
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@ -110,8 +110,11 @@ static int ac3_decode_frame(AVCodecContext *avctx,
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s->channels = ac3_channels[s->flags & 7];
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if (s->flags & AC3_LFE)
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s->channels++;
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if (s->channels < avctx->channels) {
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fprintf(stderr, "Source channels are less than specified: output to %d channels..\n", s->channels);
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if (avctx->channels == 0)
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/* No specific number of channel requested */
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avctx->channels = s->channels;
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else if (s->channels < avctx->channels) {
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fprintf(stderr, "libav: AC3 Source channels are less than specified: output to %d channels..\n", s->channels);
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avctx->channels = s->channels;
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}
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avctx->bit_rate = bit_rate;
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@ -127,14 +130,12 @@ static int ac3_decode_frame(AVCodecContext *avctx,
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s->inbuf_ptr += len;
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buf_size -= len;
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} else {
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#if 0
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flags = s->flags;
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if (avctx->channels == 1)
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flags = AC3_MONO;
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else
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else if (avctx->channels == 2)
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flags = AC3_STEREO;
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#else
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flags = s->flags;
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#endif
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else
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flags |= AC3_ADJUST_LEVEL;
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level = 1;
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if (ac3_frame (&s->state, s->inbuf, &flags, &level, 384)) {
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@ -219,6 +219,7 @@ void avcodec_string(char *buf, int buf_size, AVCodecContext *enc, int encode)
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const char *codec_name;
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AVCodec *p;
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char buf1[32];
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char *channels_str=NULL;
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int bitrate;
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if (encode)
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@ -269,12 +270,27 @@ void avcodec_string(char *buf, int buf_size, AVCodecContext *enc, int encode)
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snprintf(buf, buf_size,
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"Audio: %s",
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codec_name);
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switch (enc->channels) {
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case 1:
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channels_str = "mono";
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break;
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case 2:
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channels_str = "stereo";
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break;
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case 6:
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channels_str = "5:1";
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break;
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default:
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sprintf(channels_str, "%d channels", enc->channels);
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break;
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}
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if (enc->sample_rate) {
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snprintf(buf + strlen(buf), buf_size - strlen(buf),
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", %d Hz, %s",
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enc->sample_rate,
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enc->channels == 2 ? "stereo" : "mono");
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channels_str);
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}
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/* for PCM codecs, compute bitrate directly */
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switch(enc->codec_id) {
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case CODEC_ID_PCM_S16LE:
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