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| author | Max Kellermann <max@duempel.org> | 2013-08-03 21:00:50 +0200 |
|---|---|---|
| committer | Max Kellermann <max@duempel.org> | 2013-08-03 21:37:56 +0200 |
| commit | d1e7b4e38136f9342aad76c685a13adf0e69f869 (patch) | |
| tree | 49643b937ddfe735511b566a71398da5a945d7aa /src/decoder/SndfileDecoderPlugin.cxx | |
| parent | 67f591a9ce60651da41afc499bd9a22e25314e35 (diff) | |
| download | mpd-d1e7b4e38136f9342aad76c685a13adf0e69f869.tar.gz mpd-d1e7b4e38136f9342aad76c685a13adf0e69f869.tar.xz mpd-d1e7b4e38136f9342aad76c685a13adf0e69f869.zip | |
audio_format: convert to C++
Diffstat (limited to 'src/decoder/SndfileDecoderPlugin.cxx')
| -rw-r--r-- | src/decoder/SndfileDecoderPlugin.cxx | 14 |
1 files changed, 7 insertions, 7 deletions
diff --git a/src/decoder/SndfileDecoderPlugin.cxx b/src/decoder/SndfileDecoderPlugin.cxx index b1bb97538..63401a47b 100644 --- a/src/decoder/SndfileDecoderPlugin.cxx +++ b/src/decoder/SndfileDecoderPlugin.cxx @@ -99,7 +99,7 @@ static SF_VIRTUAL_IO vio = { * Converts a frame number to a timestamp (in seconds). */ static float -frame_to_time(sf_count_t frame, const struct audio_format *audio_format) +frame_to_time(sf_count_t frame, const AudioFormat *audio_format) { return (float)frame / (float)audio_format->sample_rate; } @@ -108,7 +108,7 @@ frame_to_time(sf_count_t frame, const struct audio_format *audio_format) * Converts a timestamp (in seconds) to a frame number. */ static sf_count_t -time_to_frame(float t, const struct audio_format *audio_format) +time_to_frame(float t, const AudioFormat *audio_format) { return (sf_count_t)(t * audio_format->sample_rate); } @@ -119,7 +119,6 @@ sndfile_stream_decode(struct decoder *decoder, struct input_stream *is) GError *error = nullptr; SNDFILE *sf; SF_INFO info; - struct audio_format audio_format; size_t frame_size; sf_count_t read_frames, num_frames; int buffer[4096]; @@ -136,18 +135,19 @@ sndfile_stream_decode(struct decoder *decoder, struct input_stream *is) /* for now, always read 32 bit samples. Later, we could lower MPD's CPU usage by reading 16 bit samples with sf_readf_short() on low-quality source files. */ - if (!audio_format_init_checked(&audio_format, info.samplerate, - SAMPLE_FORMAT_S32, + AudioFormat audio_format; + if (!audio_format_init_checked(audio_format, info.samplerate, + SampleFormat::S32, info.channels, &error)) { g_warning("%s", error->message); g_error_free(error); return; } - decoder_initialized(decoder, &audio_format, info.seekable, + decoder_initialized(decoder, audio_format, info.seekable, frame_to_time(info.frames, &audio_format)); - frame_size = audio_format_frame_size(&audio_format); + frame_size = audio_format.GetFrameSize(); read_frames = sizeof(buffer) / frame_size; do { |