From 3a666702af9a57c7e8dc5e266b28eaaa5835f5e5 Mon Sep 17 00:00:00 2001 From: Max Kellermann Date: Sat, 30 Nov 2013 13:00:41 +0100 Subject: pcm/PcmConvert: add AudioFormat parameters Don't use src_format. In the middle of Convert(), the current AudioFormat has already been modified, it's now something in between src_format and dest_format. This simplifies keeping track of what remains to be done. --- src/pcm/PcmConvert.cxx | 83 +++++++++++++++++++++++++------------------------- src/pcm/PcmConvert.hxx | 23 +++++++------- 2 files changed, 53 insertions(+), 53 deletions(-) diff --git a/src/pcm/PcmConvert.cxx b/src/pcm/PcmConvert.cxx index c405fc94f..e32a4ba4e 100644 --- a/src/pcm/PcmConvert.cxx +++ b/src/pcm/PcmConvert.cxx @@ -63,10 +63,6 @@ PcmConvert::Open(AudioFormat _src_format, AudioFormat _dest_format, src_format = _src_format; dest_format = _dest_format; - is_dsd = src_format.format == SampleFormat::DSD; - if (is_dsd) - src_format.format = SampleFormat::FLOAT; - return true; } @@ -83,7 +79,7 @@ PcmConvert::Close() } inline ConstBuffer -PcmConvert::Convert16(ConstBuffer src, Error &error) +PcmConvert::Convert16(ConstBuffer src, AudioFormat format, Error &error) { const int16_t *buf; size_t len; @@ -91,34 +87,34 @@ PcmConvert::Convert16(ConstBuffer src, Error &error) assert(dest_format.format == SampleFormat::S16); buf = pcm_convert_to_16(format_buffer, dither, - src_format.format, + format.format, src.data, src.size, &len); if (buf == nullptr) { error.Format(pcm_convert_domain, "Conversion from %s to 16 bit is not implemented", - sample_format_to_string(src_format.format)); + sample_format_to_string(format.format)); return nullptr; } - if (src_format.channels != dest_format.channels) { + if (format.channels != dest_format.channels) { buf = pcm_convert_channels_16(channels_buffer, dest_format.channels, - src_format.channels, + format.channels, buf, len, &len); if (buf == nullptr) { error.Format(pcm_convert_domain, "Conversion from %u to %u channels " "is not implemented", - src_format.channels, + format.channels, dest_format.channels); return nullptr; } } - if (src_format.sample_rate != dest_format.sample_rate) { + if (format.sample_rate != dest_format.sample_rate) { buf = resampler.Resample16(dest_format.channels, - src_format.sample_rate, buf, len, + format.sample_rate, buf, len, dest_format.sample_rate, &len, error); if (buf == nullptr) @@ -129,7 +125,7 @@ PcmConvert::Convert16(ConstBuffer src, Error &error) } inline ConstBuffer -PcmConvert::Convert24(ConstBuffer src, Error &error) +PcmConvert::Convert24(ConstBuffer src, AudioFormat format, Error &error) { const int32_t *buf; size_t len; @@ -137,33 +133,33 @@ PcmConvert::Convert24(ConstBuffer src, Error &error) assert(dest_format.format == SampleFormat::S24_P32); buf = pcm_convert_to_24(format_buffer, - src_format.format, + format.format, src.data, src.size, &len); if (buf == nullptr) { error.Format(pcm_convert_domain, "Conversion from %s to 24 bit is not implemented", - sample_format_to_string(src_format.format)); + sample_format_to_string(format.format)); return nullptr; } - if (src_format.channels != dest_format.channels) { + if (format.channels != dest_format.channels) { buf = pcm_convert_channels_24(channels_buffer, dest_format.channels, - src_format.channels, + format.channels, buf, len, &len); if (buf == nullptr) { error.Format(pcm_convert_domain, "Conversion from %u to %u channels " "is not implemented", - src_format.channels, + format.channels, dest_format.channels); return nullptr; } } - if (src_format.sample_rate != dest_format.sample_rate) { + if (format.sample_rate != dest_format.sample_rate) { buf = resampler.Resample24(dest_format.channels, - src_format.sample_rate, buf, len, + format.sample_rate, buf, len, dest_format.sample_rate, &len, error); if (buf == nullptr) @@ -174,7 +170,7 @@ PcmConvert::Convert24(ConstBuffer src, Error &error) } inline ConstBuffer -PcmConvert::Convert32(ConstBuffer src, Error &error) +PcmConvert::Convert32(ConstBuffer src, AudioFormat format, Error &error) { const int32_t *buf; size_t len; @@ -182,33 +178,33 @@ PcmConvert::Convert32(ConstBuffer src, Error &error) assert(dest_format.format == SampleFormat::S32); buf = pcm_convert_to_32(format_buffer, - src_format.format, + format.format, src.data, src.size, &len); if (buf == nullptr) { error.Format(pcm_convert_domain, "Conversion from %s to 32 bit is not implemented", - sample_format_to_string(src_format.format)); + sample_format_to_string(format.format)); return nullptr; } - if (src_format.channels != dest_format.channels) { + if (format.channels != dest_format.channels) { buf = pcm_convert_channels_32(channels_buffer, dest_format.channels, - src_format.channels, + format.channels, buf, len, &len); if (buf == nullptr) { error.Format(pcm_convert_domain, "Conversion from %u to %u channels " "is not implemented", - src_format.channels, + format.channels, dest_format.channels); return nullptr; } } - if (src_format.sample_rate != dest_format.sample_rate) { + if (format.sample_rate != dest_format.sample_rate) { buf = resampler.Resample32(dest_format.channels, - src_format.sample_rate, buf, len, + format.sample_rate, buf, len, dest_format.sample_rate, &len, error); if (buf == nullptr) @@ -219,7 +215,8 @@ PcmConvert::Convert32(ConstBuffer src, Error &error) } inline ConstBuffer -PcmConvert::ConvertFloat(ConstBuffer src, Error &error) +PcmConvert::ConvertFloat(ConstBuffer src, AudioFormat format, + Error &error) { assert(dest_format.format == SampleFormat::FLOAT); @@ -227,27 +224,27 @@ PcmConvert::ConvertFloat(ConstBuffer src, Error &error) size_t size; const float *buffer = pcm_convert_to_float(format_buffer, - src_format.format, + format.format, src.data, src.size, &size); if (buffer == nullptr) { error.Format(pcm_convert_domain, "Conversion from %s to float is not implemented", - sample_format_to_string(src_format.format)); + sample_format_to_string(format.format)); return nullptr; } /* convert channels */ - if (src_format.channels != dest_format.channels) { + if (format.channels != dest_format.channels) { buffer = pcm_convert_channels_float(channels_buffer, dest_format.channels, - src_format.channels, + format.channels, buffer, size, &size); if (buffer == nullptr) { error.Format(pcm_convert_domain, "Conversion from %u to %u channels " "is not implemented", - src_format.channels, + format.channels, dest_format.channels); return nullptr; } @@ -256,9 +253,9 @@ PcmConvert::ConvertFloat(ConstBuffer src, Error &error) /* resample with float, because this is the best format for libsamplerate */ - if (src_format.sample_rate != dest_format.sample_rate) { + if (format.sample_rate != dest_format.sample_rate) { buffer = resampler.ResampleFloat(dest_format.channels, - src_format.sample_rate, + format.sample_rate, buffer, size, dest_format.sample_rate, &size, error); @@ -275,10 +272,11 @@ PcmConvert::Convert(const void *src, size_t src_size, Error &error) { ConstBuffer buffer(src, src_size); + AudioFormat format = src_format; - if (is_dsd) { + if (format.format == SampleFormat::DSD) { auto s = ConstBuffer::FromVoid(buffer); - auto d = dsd.ToFloat(src_format.channels, + auto d = dsd.ToFloat(format.channels, false, s); if (d.IsNull()) { error.Set(pcm_convert_domain, @@ -287,23 +285,24 @@ PcmConvert::Convert(const void *src, size_t src_size, } buffer = d.ToVoid(); + format.format = SampleFormat::FLOAT; } switch (dest_format.format) { case SampleFormat::S16: - buffer = Convert16(buffer, error).ToVoid(); + buffer = Convert16(buffer, format, error).ToVoid(); break; case SampleFormat::S24_P32: - buffer = Convert24(buffer, error).ToVoid(); + buffer = Convert24(buffer, format, error).ToVoid(); break; case SampleFormat::S32: - buffer = Convert32(buffer, error).ToVoid(); + buffer = Convert32(buffer, format, error).ToVoid(); break; case SampleFormat::FLOAT: - buffer = ConvertFloat(buffer, error).ToVoid(); + buffer = ConvertFloat(buffer, format, error).ToVoid(); break; default: diff --git a/src/pcm/PcmConvert.hxx b/src/pcm/PcmConvert.hxx index 74f4e350c..12c4b26f3 100644 --- a/src/pcm/PcmConvert.hxx +++ b/src/pcm/PcmConvert.hxx @@ -52,13 +52,6 @@ class PcmConvert { AudioFormat src_format, dest_format; - /** - * Do we get DSD source data? Then this flag is true and - * src_format.format is set to SampleFormat::FLOAT, because - * the #PcmDsd class will convert it to floating point. - */ - bool is_dsd; - public: PcmConvert(); ~PcmConvert(); @@ -92,10 +85,18 @@ public: Error &error); private: - ConstBuffer Convert16(ConstBuffer src, Error &error); - ConstBuffer Convert24(ConstBuffer src, Error &error); - ConstBuffer Convert32(ConstBuffer src, Error &error); - ConstBuffer ConvertFloat(ConstBuffer src, Error &error); + ConstBuffer Convert16(ConstBuffer src, + AudioFormat format, + Error &error); + ConstBuffer Convert24(ConstBuffer src, + AudioFormat format, + Error &error); + ConstBuffer Convert32(ConstBuffer src, + AudioFormat format, + Error &error); + ConstBuffer ConvertFloat(ConstBuffer src, + AudioFormat format, + Error &error); }; extern const Domain pcm_convert_domain; -- cgit v1.2.3