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decoder/ffmpeg: use avcodec_decode_audio4(), support libavcodec 0.8

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This commit is contained in:
Max Kellermann 2012-01-04 22:10:38 +01:00
parent 531948358b
commit 4e6bc77a70
2 changed files with 59 additions and 2 deletions
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2
NEWS
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@ -2,7 +2,7 @@ ver 0.16.7 (2011/??/??)
* input:
- ffmpeg: support libavformat 0.7
* decoder:
- ffmpeg: support libavformat 0.7
- ffmpeg: support libavformat 0.7, libavcodec 0.8
* output:
- httpd: fix excessive buffering
- openal: force 16 bit playback, as 8 bit doesn't work

59
src/decoder/ffmpeg_decoder_plugin.c
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@ -208,6 +208,7 @@ ffmpeg_find_audio_stream(const AVFormatContext *format_context)
return -1;
}
#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(53,25,0)
/**
* On some platforms, libavcodec wants the output buffer aligned to 16
* bytes (because it uses SSE/Altivec internally). This function
@ -222,6 +223,7 @@ align16(void *p, size_t *length_p)
*length_p -= add;
return (char *)p + add;
}
#endif
G_GNUC_CONST
static double
@ -241,6 +243,40 @@ time_to_ffmpeg(double t, const AVRational time_base)
time_base);
}
#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(53,25,0)
/**
* Copy PCM data from a AVFrame to an interleaved buffer.
*/
static int
copy_interleave_frame(const AVCodecContext *codec_context,
const AVFrame *frame,
uint8_t *buffer, size_t buffer_size)
{
int plane_size;
const int data_size =
av_samples_get_buffer_size(&plane_size,
codec_context->channels,
frame->nb_samples,
codec_context->sample_fmt, 1);
if (buffer_size < (size_t)data_size)
/* buffer is too small - shouldn't happen */
return AVERROR(EINVAL);
if (av_sample_fmt_is_planar(codec_context->sample_fmt) &&
codec_context->channels > 1) {
for (int i = 0, channels = codec_context->channels;
i < channels; i++) {
memcpy(buffer, frame->extended_data[i], plane_size);
buffer += plane_size;
}
} else {
memcpy(buffer, frame->extended_data[0], data_size);
}
return data_size;
}
#endif
static enum decoder_command
ffmpeg_send_packet(struct decoder *decoder, struct input_stream *is,
const AVPacket *packet,
@ -258,9 +294,15 @@ ffmpeg_send_packet(struct decoder *decoder, struct input_stream *is,
int packet_size = packet->size;
#endif
#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(53,25,0)
uint8_t aligned_buffer[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2 + 16];
const size_t buffer_size = sizeof(aligned_buffer);
#else
/* libavcodec < 0.8 needs an aligned buffer */
uint8_t audio_buf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2 + 16];
size_t buffer_size = sizeof(audio_buf);
int16_t *aligned_buffer = align16(audio_buf, &buffer_size);
#endif
enum decoder_command cmd = DECODE_COMMAND_NONE;
while (
@ -271,7 +313,22 @@ ffmpeg_send_packet(struct decoder *decoder, struct input_stream *is,
#endif
cmd == DECODE_COMMAND_NONE) {
int audio_size = buffer_size;
#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(52,25,0)
#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(53,25,0)
AVFrame frame;
int got_frame = 0;
int len = avcodec_decode_audio4(codec_context,
&frame, &got_frame,
&packet2);
if (len >= 0 && got_frame) {
audio_size = copy_interleave_frame(codec_context,
&frame,
aligned_buffer,
buffer_size);
if (audio_size < 0)
len = audio_size;
} else if (len >= 0)
len = -1;
#elif LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(52,25,0)
int len = avcodec_decode_audio3(codec_context,
aligned_buffer, &audio_size,
&packet2);