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https://github.com/FFmpeg/FFmpeg.git
synced 2025-01-08 13:22:53 +02:00
parse sample rate instead of setting a default one
Originally committed as revision 10799 to svn://svn.ffmpeg.org/ffmpeg/trunk
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parent
fd402a5a63
commit
6c867e0463
@ -35,7 +35,6 @@
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#define MV0K_TAG MKTAG('M', 'V', '0', 'K')
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#define MV0F_TAG MKTAG('M', 'V', '0', 'F')
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#define EA_SAMPLE_RATE 22050
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#define EA_BITS_PER_SAMPLE 16
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#define EA_PREAMBLE_SIZE 8
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@ -52,6 +51,7 @@ typedef struct EaDemuxContext {
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int64_t audio_pts;
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int sample_rate;
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int num_channels;
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int num_samples;
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} EaDemuxContext;
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@ -82,8 +82,9 @@ static int process_audio_header_elements(AVFormatContext *s)
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int inHeader = 1;
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EaDemuxContext *ea = s->priv_data;
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ByteIOContext *pb = &s->pb;
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int compression_type = -1;
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int compression_type = -1, revision = -1;
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ea->sample_rate = -1;
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ea->num_channels = 1;
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while (inHeader) {
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@ -100,6 +101,10 @@ static int process_audio_header_elements(AVFormatContext *s)
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subbyte = get_byte(pb);
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switch (subbyte) {
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case 0x80:
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revision = read_arbitary(pb);
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av_log (s, AV_LOG_INFO, "revision (element 0x80) set to 0x%08x\n", revision);
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break;
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case 0x82:
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ea->num_channels = read_arbitary(pb);
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av_log (s, AV_LOG_INFO, "num_channels (element 0x82) set to 0x%08x\n", ea->num_channels);
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@ -108,6 +113,10 @@ static int process_audio_header_elements(AVFormatContext *s)
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compression_type = read_arbitary(pb);
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av_log (s, AV_LOG_INFO, "compression_type (element 0x83) set to 0x%08x\n", compression_type);
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break;
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case 0x84:
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ea->sample_rate = read_arbitary(pb);
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av_log (s, AV_LOG_INFO, "sample_rate (element 0x84) set to %i\n", ea->sample_rate);
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break;
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case 0x85:
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ea->num_samples = read_arbitary(pb);
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av_log (s, AV_LOG_INFO, "num_samples (element 0x85) set to 0x%08x\n", ea->num_samples);
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@ -146,6 +155,9 @@ static int process_audio_header_elements(AVFormatContext *s)
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return 0;
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}
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if (ea->sample_rate == -1)
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ea->sample_rate = revision==3 ? 48000 : 22050;
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return 1;
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}
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@ -250,12 +262,12 @@ static int ea_read_header(AVFormatContext *s,
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st = av_new_stream(s, 0);
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if (!st)
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return AVERROR(ENOMEM);
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av_set_pts_info(st, 33, 1, EA_SAMPLE_RATE);
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av_set_pts_info(st, 33, 1, ea->sample_rate);
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st->codec->codec_type = CODEC_TYPE_AUDIO;
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st->codec->codec_id = ea->audio_codec;
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st->codec->codec_tag = 0; /* no tag */
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st->codec->channels = ea->num_channels;
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st->codec->sample_rate = EA_SAMPLE_RATE;
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st->codec->sample_rate = ea->sample_rate;
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st->codec->bits_per_sample = EA_BITS_PER_SAMPLE;
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st->codec->bit_rate = st->codec->channels * st->codec->sample_rate *
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st->codec->bits_per_sample / 4;
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@ -295,7 +307,7 @@ static int ea_read_packet(AVFormatContext *s,
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pkt->stream_index = ea->audio_stream_index;
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pkt->pts = 90000;
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pkt->pts *= ea->audio_frame_counter;
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pkt->pts /= EA_SAMPLE_RATE;
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pkt->pts /= ea->sample_rate;
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/* 2 samples/byte, 1 or 2 samples per frame depending
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* on stereo; chunk also has 12-byte header */
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