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faad: faad_decoder_init() returns an audio_format

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Instead of returning the sample rate and channel count as separate
values, fill an audio_format struct.
This commit is contained in:
Max Kellermann 2009-02-17 23:35:49 +01:00
parent 161bfc4bc0
commit 76b0601f6b
1 changed files with 23 additions and 25 deletions
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48
src/decoder/faad_plugin.c
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@ -223,7 +223,7 @@ faad_song_duration(struct decoder_buffer *buffer, struct input_stream *is)
*/
static bool
faad_decoder_init(faacDecHandle decoder, struct decoder_buffer *buffer,
uint32_t *sample_rate, uint8_t *channels)
struct audio_format *audio_format)
{
union {
/* deconst hack for libfaad */
@ -232,13 +232,15 @@ faad_decoder_init(faacDecHandle decoder, struct decoder_buffer *buffer,
} u;
size_t length;
int32_t nbytes;
uint32_t sample_rate;
uint8_t channels;
#ifdef HAVE_FAAD_LONG
/* neaacdec.h declares all arguments as "unsigned long", but
internally expects uint32_t pointers. To avoid gcc
warnings, use this workaround. */
unsigned long *sample_rate_r = (unsigned long *)(void *)sample_rate;
unsigned long *sample_rate_r = (unsigned long *)(void *)&sample_rate;
#else
uint32_t *sample_rate_r = sample_rate;
uint32_t *sample_rate_r = &sample_rate;
#endif
u.in = decoder_buffer_read(buffer, &length);
@ -249,11 +251,18 @@ faad_decoder_init(faacDecHandle decoder, struct decoder_buffer *buffer,
#ifdef HAVE_FAAD_BUFLEN_FUNCS
length,
#endif
sample_rate_r, channels);
sample_rate_r, &channels);
if (nbytes < 0)
return false;
decoder_buffer_consume(buffer, nbytes);
*audio_format = (struct audio_format){
.bits = 16,
.channels = channels,
.sample_rate = sample_rate,
};
return true;
}
@ -294,8 +303,6 @@ faad_get_file_time_float(const char *file)
float length;
faacDecHandle decoder;
faacDecConfigurationPtr config;
uint32_t sample_rate;
unsigned char channels;
struct input_stream is;
if (!input_stream_open(&is, file))
@ -307,6 +314,7 @@ faad_get_file_time_float(const char *file)
if (length < 0) {
bool ret;
struct audio_format audio_format;
decoder = faacDecOpen();
@ -316,9 +324,8 @@ faad_get_file_time_float(const char *file)
decoder_buffer_fill(buffer);
ret = faad_decoder_init(decoder, buffer,
&sample_rate, &channels);
if (ret && sample_rate > 0 && channels > 0)
ret = faad_decoder_init(decoder, buffer, &audio_format);
if (ret && audio_format_valid(&audio_format))
length = 0;
faacDecClose(decoder);
@ -351,8 +358,6 @@ faad_stream_decode(struct decoder *mpd_decoder, struct input_stream *is)
struct audio_format audio_format;
faacDecFrameInfo frame_info;
faacDecConfigurationPtr config;
uint32_t sample_rate;
unsigned char channels;
unsigned long sample_count;
bool ret;
const void *decoded;
@ -384,20 +389,13 @@ faad_stream_decode(struct decoder *mpd_decoder, struct input_stream *is)
decoder_buffer_fill(buffer);
}
ret = faad_decoder_init(decoder, buffer,
&sample_rate, &channels);
ret = faad_decoder_init(decoder, buffer, &audio_format);
if (!ret) {
g_warning("Error not a AAC stream.\n");
faacDecClose(decoder);
return;
}
audio_format = (struct audio_format){
.bits = 16,
.channels = channels,
.sample_rate = sample_rate,
};
if (!audio_format_valid(&audio_format)) {
g_warning("invalid audio format\n");
faacDecClose(decoder);
@ -421,16 +419,16 @@ faad_stream_decode(struct decoder *mpd_decoder, struct input_stream *is)
break;
}
if (frame_info.channels != channels) {
if (frame_info.channels != audio_format.channels) {
g_warning("channel count changed from %u to %u",
channels, frame_info.channels);
audio_format.channels, frame_info.channels);
break;
}
#ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE
if (frame_info.samplerate != sample_rate) {
if (frame_info.samplerate != audio_format.sample_rate) {
g_warning("sample rate changed from %u to %lu",
sample_rate,
audio_format.sample_rate,
(unsigned long)frame_info.samplerate);
break;
}
@ -441,11 +439,11 @@ faad_stream_decode(struct decoder *mpd_decoder, struct input_stream *is)
sample_count = (unsigned long)frame_info.samples;
if (sample_count > 0) {
bit_rate = frame_info.bytesconsumed * 8.0 *
frame_info.channels * sample_rate /
frame_info.channels * audio_format.sample_rate /
frame_info.samples / 1000 + 0.5;
file_time +=
(float)(frame_info.samples) / frame_info.channels /
sample_rate;
audio_format.sample_rate;
}
decoded_length = sample_count * 2;