diff options
| -rw-r--r-- | src/decoder/faad_plugin.c | 48 |
1 files changed, 23 insertions, 25 deletions
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; |