From e9127523db55a267f67532fd61e913f2879324fc Mon Sep 17 00:00:00 2001 From: Max Kellermann Date: Sat, 30 Nov 2013 14:10:31 +0100 Subject: pcm/PcmConvert: move code to new class GluePcmResampler --- src/pcm/GlueResampler.cxx | 94 ++++++++++++++++++++++++++++++++ src/pcm/GlueResampler.hxx | 44 +++++++++++++++ src/pcm/PcmConvert.cxx | 134 +++++----------------------------------------- src/pcm/PcmConvert.hxx | 19 +------ 4 files changed, 152 insertions(+), 139 deletions(-) create mode 100644 src/pcm/GlueResampler.cxx create mode 100644 src/pcm/GlueResampler.hxx (limited to 'src/pcm') diff --git a/src/pcm/GlueResampler.cxx b/src/pcm/GlueResampler.cxx new file mode 100644 index 000000000..8aee98f38 --- /dev/null +++ b/src/pcm/GlueResampler.cxx @@ -0,0 +1,94 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include "config.h" +#include "GlueResampler.hxx" +#include "PcmConvert.hxx" +#include "PcmFormat.hxx" +#include "util/ConstBuffer.hxx" +#include "util/Error.hxx" + +bool +GluePcmResampler::Open(AudioFormat _src_format, unsigned _new_sample_rate, + gcc_unused Error &error) +{ + src_format = _src_format; + new_sample_rate = _new_sample_rate; + + return true; +} + +void +GluePcmResampler::Close() +{ + resampler.Reset(); +} + +ConstBuffer +GluePcmResampler::Resample(ConstBuffer src, Error &error) +{ + const void *result; + size_t size; + + switch (src_format.format) { + case SampleFormat::S16: + result = resampler.Resample16(src_format.channels, + src_format.sample_rate, + (const int16_t *)src.data, + src.size, + new_sample_rate, &size, + error); + break; + + case SampleFormat::S24_P32: + result = resampler.Resample24(src_format.channels, + src_format.sample_rate, + (const int32_t *)src.data, + src.size, + new_sample_rate, &size, + error); + break; + + case SampleFormat::S32: + result = resampler.Resample24(src_format.channels, + src_format.sample_rate, + (const int32_t *)src.data, + src.size, + new_sample_rate, &size, + error); + break; + + case SampleFormat::FLOAT: + result = resampler.ResampleFloat(src_format.channels, + src_format.sample_rate, + (const float *)src.data, + src.size, + new_sample_rate, &size, + error); + break; + + default: + error.Format(pcm_convert_domain, + "Resampling %s is not implemented", + sample_format_to_string(src_format.format)); + return nullptr; + } + + return { result, size }; +} diff --git a/src/pcm/GlueResampler.hxx b/src/pcm/GlueResampler.hxx new file mode 100644 index 000000000..a7e0a84f2 --- /dev/null +++ b/src/pcm/GlueResampler.hxx @@ -0,0 +1,44 @@ +/* + * Copyright (C) 2003-2013 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#ifndef MPD_GLUE_RESAMPLER_HXX +#define MPD_GLUE_RESAMPLER_HXX + +#include "check.h" +#include "AudioFormat.hxx" +#include "PcmResample.hxx" + +class Error; +template struct ConstBuffer; + +class GluePcmResampler { + PcmResampler resampler; + + AudioFormat src_format; + unsigned new_sample_rate; + +public: + bool Open(AudioFormat src_format, unsigned new_sample_rate, + Error &error); + void Close(); + + ConstBuffer Resample(ConstBuffer src, Error &error); +}; + +#endif diff --git a/src/pcm/PcmConvert.cxx b/src/pcm/PcmConvert.cxx index bc289460d..0552962aa 100644 --- a/src/pcm/PcmConvert.cxx +++ b/src/pcm/PcmConvert.cxx @@ -78,6 +78,10 @@ PcmConvert::Open(AudioFormat _src_format, AudioFormat _dest_format, return false; } + if (format.sample_rate != dest_format.sample_rate && + !resampler.Open(format, dest_format.sample_rate, error)) + return false; + return true; } @@ -91,7 +95,9 @@ PcmConvert::Close() format_converter.Close(); dsd.Reset(); - resampler.Reset(); + + if (src_format.sample_rate != dest_format.sample_rate) + resampler.Close(); #ifndef NDEBUG src_format.Clear(); @@ -99,102 +105,6 @@ PcmConvert::Close() #endif } -inline ConstBuffer -PcmConvert::Convert16(ConstBuffer src, AudioFormat format, - Error &error) -{ - assert(format.format == SampleFormat::S16); - assert(dest_format.format == SampleFormat::S16); - assert(format.channels == dest_format.channels); - - auto buf = src.data; - size_t len = src.size * sizeof(*src.data); - - if (format.sample_rate != dest_format.sample_rate) { - buf = resampler.Resample16(dest_format.channels, - format.sample_rate, buf, len, - dest_format.sample_rate, &len, - error); - if (buf == nullptr) - return nullptr; - } - - return ConstBuffer::FromVoid({buf, len}); -} - -inline ConstBuffer -PcmConvert::Convert24(ConstBuffer src, AudioFormat format, - Error &error) -{ - assert(format.format == SampleFormat::S24_P32); - assert(dest_format.format == SampleFormat::S24_P32); - assert(format.channels == dest_format.channels); - - auto buf = src.data; - size_t len = src.size * sizeof(*src.data); - - if (format.sample_rate != dest_format.sample_rate) { - buf = resampler.Resample24(dest_format.channels, - format.sample_rate, buf, len, - dest_format.sample_rate, &len, - error); - if (buf == nullptr) - return nullptr; - } - - return ConstBuffer::FromVoid({buf, len}); -} - -inline ConstBuffer -PcmConvert::Convert32(ConstBuffer src, AudioFormat format, - Error &error) -{ - assert(format.format == SampleFormat::S32); - assert(dest_format.format == SampleFormat::S32); - assert(format.channels == dest_format.channels); - - auto buf = src.data; - size_t len = src.size * sizeof(*src.data); - - if (format.sample_rate != dest_format.sample_rate) { - buf = resampler.Resample32(dest_format.channels, - format.sample_rate, buf, len, - dest_format.sample_rate, &len, - error); - if (buf == nullptr) - return nullptr; - } - - return ConstBuffer::FromVoid({buf, len}); -} - -inline ConstBuffer -PcmConvert::ConvertFloat(ConstBuffer src, AudioFormat format, - Error &error) -{ - assert(format.format == SampleFormat::FLOAT); - assert(dest_format.format == SampleFormat::FLOAT); - assert(format.channels == dest_format.channels); - - auto buffer = src.data; - size_t size = src.size * sizeof(*src.data); - - /* resample with float, because this is the best format for - libsamplerate */ - - if (format.sample_rate != dest_format.sample_rate) { - buffer = resampler.ResampleFloat(dest_format.channels, - format.sample_rate, - buffer, size, - dest_format.sample_rate, - &size, error); - if (buffer == nullptr) - return nullptr; - } - - return ConstBuffer::FromVoid({buffer, size}); -} - const void * PcmConvert::Convert(const void *src, size_t src_size, size_t *dest_size_r, @@ -233,32 +143,12 @@ PcmConvert::Convert(const void *src, size_t src_size, format.channels = dest_format.channels; } - switch (dest_format.format) { - case SampleFormat::S16: - buffer = Convert16(ConstBuffer::FromVoid(buffer), - format, error).ToVoid(); - break; - - case SampleFormat::S24_P32: - buffer = Convert24(ConstBuffer::FromVoid(buffer), - format, error).ToVoid(); - break; - - case SampleFormat::S32: - buffer = Convert32(ConstBuffer::FromVoid(buffer), - format, error).ToVoid(); - break; - - case SampleFormat::FLOAT: - buffer = ConvertFloat(ConstBuffer::FromVoid(buffer), - format, error).ToVoid(); - break; + if (format.sample_rate != dest_format.sample_rate) { + buffer = resampler.Resample(buffer, error); + if (buffer.IsNull()) + return nullptr; - default: - error.Format(pcm_convert_domain, - "PCM conversion to %s is not implemented", - sample_format_to_string(dest_format.format)); - return nullptr; + format.sample_rate = dest_format.sample_rate; } *dest_size_r = buffer.size; diff --git a/src/pcm/PcmConvert.hxx b/src/pcm/PcmConvert.hxx index d6e113915..9835045d6 100644 --- a/src/pcm/PcmConvert.hxx +++ b/src/pcm/PcmConvert.hxx @@ -21,10 +21,10 @@ #define PCM_CONVERT_HXX #include "PcmDsd.hxx" -#include "PcmResample.hxx" #include "PcmBuffer.hxx" #include "FormatConverter.hxx" #include "ChannelsConverter.hxx" +#include "GlueResampler.hxx" #include "AudioFormat.hxx" #include @@ -41,10 +41,9 @@ class Domain; class PcmConvert { PcmDsd dsd; - PcmResampler resampler; - PcmFormatConverter format_converter; PcmChannelsConverter channels_converter; + GluePcmResampler resampler; AudioFormat src_format, dest_format; @@ -79,20 +78,6 @@ public: const void *Convert(const void *src, size_t src_size, size_t *dest_size_r, Error &error); - -private: - 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