From 76b0601f6b0464f446414430ce7351e097229f58 Mon Sep 17 00:00:00 2001
From: Max Kellermann <max@duempel.org>
Date: Tue, 17 Feb 2009 23:35:49 +0100
Subject: faad: faad_decoder_init() returns an audio_format
Instead of returning the sample rate and channel count as separate
values, fill an audio_format struct.
---
src/decoder/faad_plugin.c | 48 +++++++++++++++++++++++------------------------
1 file changed, 23 insertions(+), 25 deletions(-)
(limited to 'src')
diff --git a/src/decoder/faad_plugin.c b/src/decoder/faad_plugin.c
index 290a7c7f0..96a6bd0d4 100644
--- a/src/decoder/faad_plugin.c
+++ b/src/decoder/faad_plugin.c
@@ -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;
--
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