feat: add OGG support for readSamplesFrom* methods

.vscode/launch.json

@@ -8,7 +8,7 @@
             "name": "Flutter debug",
             "type": "dart",
             "request": "launch",
-            "program": "lib/audio_context/audio_context.dart",
+            "program": "lib/wave_data/wave_data.dart",
             "flutterMode": "debug",
             // "env": {
             //     "NO_OPUS_OGG_LIBS": "1"

.vscode/tasks.json

@@ -9,7 +9,7 @@
         },
         {
             "label": "compile linux debug",
-            "command": "cd ${workspaceFolder}/example; flutter build linux -t lib/buffer_stream/simple_noise_stream.dart --debug",
+            "command": "cd ${workspaceFolder}/example; flutter build linux -t lib/wave_data/wave_data.dart --debug",
             "type": "shell"
         },
         {
@@ -29,7 +29,7 @@
         },
         {
             "label": "compile web wasm release",
-            "command": "cd ${workspaceFolder}/example; flutter run -d chrome --wasm --web-browser-flag '--disable-web-security' -t lib/buffer_stream/simple_noise_stream.dart --release",
+            "command": "cd ${workspaceFolder}/example; flutter run -d chrome --wasm --web-browser-flag '--disable-web-security' -t lib/wave_data/wave_data.dart --release",
             "type": "shell"
         },
         {

android/CMakeLists.txt

@@ -41,6 +41,7 @@ list(APPEND PLUGIN_SOURCES
   "${SRC_DIR}/analyzer.cpp"
   "${SRC_DIR}/synth/basic_wave.cpp"
   "${SRC_DIR}/waveform/waveform.cpp"
+  "${SRC_DIR}/waveform/miniaudio_libvorbis.cpp"
   "${SRC_DIR}/audiobuffer/audiobuffer.cpp"
   "${SRC_DIR}/filters/filters.cpp"
   "${SRC_DIR}/filters/pitch_shift_filter.cpp"
@@ -82,6 +83,12 @@ if(NOT DEFINED ENV{NO_OPUS_OGG_LIBS})
     target_link_libraries(${PLUGIN_NAME} PRIVATE ${OGG_LIBRARY})
     set(INSTALL_BUNDLE_LIB_DIR "${CMAKE_INSTALL_PREFIX}/lib")
 
+    # Add vorbis library
+    set(VORBIS_LIBRARY ${CMAKE_CURRENT_SOURCE_DIR}/libs/${ANDROID_ABI}/libvorbis.so)
+    target_link_libraries(${PLUGIN_NAME} PRIVATE ${VORBIS_LIBRARY})
+    set(VORBISFILE_LIBRARY ${CMAKE_CURRENT_SOURCE_DIR}/libs/${ANDROID_ABI}/libvorbisfile.so)
+    target_link_libraries(${PLUGIN_NAME} PRIVATE ${VORBISFILE_LIBRARY})
+
     include_directories(${CMAKE_CURRENT_SOURCE_DIR}/include)
 else()
     message("NO_OPUS_OGG_LIBS has been set. Not linking Opus and Ogg libraries!")

android/include/opus/opus.h

@@ -268,6 +268,42 @@ OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_encode(
     opus_int32 max_data_bytes
 ) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
 
+/** Encodes an Opus frame.
+  * @param [in] st <tt>OpusEncoder*</tt>: Encoder state
+  * @param [in] pcm <tt>opus_int32*</tt>: Input signal (interleaved if 2 channels) representing (or slightly exceeding) 24-bit values. length is frame_size*channels*sizeof(opus_int32)
+  * @param [in] frame_size <tt>int</tt>: Number of samples per channel in the
+  *                                      input signal.
+  *                                      This must be an Opus frame size for
+  *                                      the encoder's sampling rate.
+  *                                      For example, at 48 kHz the permitted
+  *                                      values are 120, 240, 480, 960, 1920,
+  *                                      and 2880.
+  *                                      Passing in a duration of less than
+  *                                      10 ms (480 samples at 48 kHz) will
+  *                                      prevent the encoder from using the LPC
+  *                                      or hybrid modes.
+  * @param [out] data <tt>unsigned char*</tt>: Output payload.
+  *                                            This must contain storage for at
+  *                                            least \a max_data_bytes.
+  * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
+  *                                                 memory for the output
+  *                                                 payload. This may be
+  *                                                 used to impose an upper limit on
+  *                                                 the instant bitrate, but should
+  *                                                 not be used as the only bitrate
+  *                                                 control. Use #OPUS_SET_BITRATE to
+  *                                                 control the bitrate.
+  * @returns The length of the encoded packet (in bytes) on success or a
+  *          negative error code (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_encode24(
+    OpusEncoder *st,
+    const opus_int32 *pcm,
+    int frame_size,
+    unsigned char *data,
+    opus_int32 max_data_bytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
 /** Encodes an Opus frame from floating point input.
   * @param [in] st <tt>OpusEncoder*</tt>: Encoder state
   * @param [in] pcm <tt>float*</tt>: Input in float format (interleaved if 2 channels), with a normal range of +/-1.0.
@@ -483,6 +519,31 @@ OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decode(
     int decode_fec
 ) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
 
+/** Decode an Opus packet.
+  * @param [in] st <tt>OpusDecoder*</tt>: Decoder state
+  * @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
+  * @param [in] len <tt>opus_int32</tt>: Number of bytes in payload*
+  * @param [out] pcm <tt>opus_int32*</tt>: Output signal (interleaved if 2 channels) representing (or slightly exceeding) 24-bit values. length
+  *  is frame_size*channels*sizeof(opus_int32)
+  * @param [in] frame_size Number of samples per channel of available space in \a pcm.
+  *  If this is less than the maximum packet duration (120ms; 5760 for 48kHz), this function will
+  *  not be capable of decoding some packets. In the case of PLC (data==NULL) or FEC (decode_fec=1),
+  *  then frame_size needs to be exactly the duration of audio that is missing, otherwise the
+  *  decoder will not be in the optimal state to decode the next incoming packet. For the PLC and
+  *  FEC cases, frame_size <b>must</b> be a multiple of 2.5 ms.
+  * @param [in] decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band forward error correction data be
+  *  decoded. If no such data is available, the frame is decoded as if it were lost.
+  * @returns Number of decoded samples or @ref opus_errorcodes
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decode24(
+    OpusDecoder *st,
+    const unsigned char *data,
+    opus_int32 len,
+    opus_int32 *pcm,
+    int frame_size,
+    int decode_fec
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
 /** Decode an Opus packet with floating point output.
   * @param [in] st <tt>OpusDecoder*</tt>: Decoder state
   * @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
@@ -596,7 +657,7 @@ OPUS_EXPORT int opus_dred_parse(OpusDREDDecoder *dred_dec, OpusDRED *dred, const
   */
 OPUS_EXPORT int opus_dred_process(OpusDREDDecoder *dred_dec, const OpusDRED *src, OpusDRED *dst);
 
-/** Decode audio from an Opus DRED packet with floating point output.
+/** Decode audio from an Opus DRED packet with 16-bit output.
   * @param [in] st <tt>OpusDecoder*</tt>: Decoder state
   * @param [in] dred <tt>OpusDRED*</tt>: DRED state
   * @param [in] dred_offset <tt>opus_int32</tt>: position of the redundancy to decode (in samples before the beginning of the real audio data in the packet).
@@ -608,6 +669,18 @@ OPUS_EXPORT int opus_dred_process(OpusDREDDecoder *dred_dec, const OpusDRED *src
   */
 OPUS_EXPORT int opus_decoder_dred_decode(OpusDecoder *st, const OpusDRED *dred, opus_int32 dred_offset, opus_int16 *pcm, opus_int32 frame_size);
 
+/** Decode audio from an Opus DRED packet with 24-bit output.
+  * @param [in] st <tt>OpusDecoder*</tt>: Decoder state
+  * @param [in] dred <tt>OpusDRED*</tt>: DRED state
+  * @param [in] dred_offset <tt>opus_int32</tt>: position of the redundancy to decode (in samples before the beginning of the real audio data in the packet).
+  * @param [out] pcm <tt>opus_int32*</tt>: Output signal (interleaved if 2 channels). length
+  *  is frame_size*channels*sizeof(opus_int16)
+  * @param [in] frame_size Number of samples per channel to decode in \a pcm.
+  *  frame_size <b>must</b> be a multiple of 2.5 ms.
+  * @returns Number of decoded samples or @ref opus_errorcodes
+  */
+OPUS_EXPORT int opus_decoder_dred_decode24(OpusDecoder *st, const OpusDRED *dred, opus_int32 dred_offset, opus_int32 *pcm, opus_int32 frame_size);
+
 /** Decode audio from an Opus DRED packet with floating point output.
   * @param [in] st <tt>OpusDecoder*</tt>: Decoder state
   * @param [in] dred <tt>OpusDRED*</tt>: DRED state

android/include/opus/opus_multistream.h

@@ -382,6 +382,44 @@ OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_encode(
     opus_int32 max_data_bytes
 ) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
 
+/** Encodes a multistream Opus frame.
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state.
+  * @param[in] pcm <tt>const opus_int32*</tt>: The input signal as interleaved
+  *                                            samples representing (or slightly exceeding) 24-bit values.
+  *                                            This must contain
+  *                                            <code>frame_size*channels</code>
+  *                                            samples.
+  * @param frame_size <tt>int</tt>: Number of samples per channel in the input
+  *                                 signal.
+  *                                 This must be an Opus frame size for the
+  *                                 encoder's sampling rate.
+  *                                 For example, at 48 kHz the permitted values
+  *                                 are 120, 240, 480, 960, 1920, and 2880.
+  *                                 Passing in a duration of less than 10 ms
+  *                                 (480 samples at 48 kHz) will prevent the
+  *                                 encoder from using the LPC or hybrid modes.
+  * @param[out] data <tt>unsigned char*</tt>: Output payload.
+  *                                           This must contain storage for at
+  *                                           least \a max_data_bytes.
+  * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
+  *                                                 memory for the output
+  *                                                 payload. This may be
+  *                                                 used to impose an upper limit on
+  *                                                 the instant bitrate, but should
+  *                                                 not be used as the only bitrate
+  *                                                 control. Use #OPUS_SET_BITRATE to
+  *                                                 control the bitrate.
+  * @returns The length of the encoded packet (in bytes) on success or a
+  *          negative error code (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_encode24(
+    OpusMSEncoder *st,
+    const opus_int32 *pcm,
+    int frame_size,
+    unsigned char *data,
+    opus_int32 max_data_bytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
 /** Encodes a multistream Opus frame from floating point input.
   * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state.
   * @param[in] pcm <tt>const float*</tt>: The input signal as interleaved
@@ -591,6 +629,44 @@ OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_decode(
     int decode_fec
 ) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
 
+/** Decode a multistream Opus packet.
+  * @param st <tt>OpusMSDecoder*</tt>: Multistream decoder state.
+  * @param[in] data <tt>const unsigned char*</tt>: Input payload.
+  *                                                Use a <code>NULL</code>
+  *                                                pointer to indicate packet
+  *                                                loss.
+  * @param len <tt>opus_int32</tt>: Number of bytes in payload.
+  * @param[out] pcm <tt>opus_int32*</tt>: Output signal, with interleaved
+  *                                       samples representing (or slightly exceeding) 24-bit values.
+  *                                       This must contain room for
+  *                                       <code>frame_size*channels</code>
+  *                                       samples.
+  * @param frame_size <tt>int</tt>: The number of samples per channel of
+  *                                 available space in \a pcm.
+  *                                 If this is less than the maximum packet duration
+  *                                 (120 ms; 5760 for 48kHz), this function will not be capable
+  *                                 of decoding some packets. In the case of PLC (data==NULL)
+  *                                 or FEC (decode_fec=1), then frame_size needs to be exactly
+  *                                 the duration of audio that is missing, otherwise the
+  *                                 decoder will not be in the optimal state to decode the
+  *                                 next incoming packet. For the PLC and FEC cases, frame_size
+  *                                 <b>must</b> be a multiple of 2.5 ms.
+  * @param decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band
+  *                                 forward error correction data be decoded.
+  *                                 If no such data is available, the frame is
+  *                                 decoded as if it were lost.
+  * @returns Number of samples decoded on success or a negative error code
+  *          (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_decode24(
+    OpusMSDecoder *st,
+    const unsigned char *data,
+    opus_int32 len,
+    opus_int32 *pcm,
+    int frame_size,
+    int decode_fec
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
 /** Decode a multistream Opus packet with floating point output.
   * @param st <tt>OpusMSDecoder*</tt>: Multistream decoder state.
   * @param[in] data <tt>const unsigned char*</tt>: Input payload.

android/include/opus/opus_projection.h

@@ -260,6 +260,44 @@ OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_projection_encode(
     opus_int32 max_data_bytes
 ) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
 
+/** Encodes a projection Opus frame.
+  * @param st <tt>OpusProjectionEncoder*</tt>: Projection encoder state.
+  * @param[in] pcm <tt>const opus_int32*</tt>: The input signal as interleaved
+  *                                            samples representing (or slightly exceeding) 24-bit values.
+  *                                            This must contain
+  *                                            <code>frame_size*channels</code>
+  *                                            samples.
+  * @param frame_size <tt>int</tt>: Number of samples per channel in the input
+  *                                 signal.
+  *                                 This must be an Opus frame size for the
+  *                                 encoder's sampling rate.
+  *                                 For example, at 48 kHz the permitted values
+  *                                 are 120, 240, 480, 960, 1920, and 2880.
+  *                                 Passing in a duration of less than 10 ms
+  *                                 (480 samples at 48 kHz) will prevent the
+  *                                 encoder from using the LPC or hybrid modes.
+  * @param[out] data <tt>unsigned char*</tt>: Output payload.
+  *                                           This must contain storage for at
+  *                                           least \a max_data_bytes.
+  * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
+  *                                                 memory for the output
+  *                                                 payload. This may be
+  *                                                 used to impose an upper limit on
+  *                                                 the instant bitrate, but should
+  *                                                 not be used as the only bitrate
+  *                                                 control. Use #OPUS_SET_BITRATE to
+  *                                                 control the bitrate.
+  * @returns The length of the encoded packet (in bytes) on success or a
+  *          negative error code (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_projection_encode24(
+    OpusProjectionEncoder *st,
+    const opus_int32 *pcm,
+    int frame_size,
+    unsigned char *data,
+    opus_int32 max_data_bytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
 
 /** Encodes a projection Opus frame from floating point input.
   * @param st <tt>OpusProjectionEncoder*</tt>: Projection encoder state.
@@ -493,6 +531,43 @@ OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_projection_decode(
     int decode_fec
 ) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
 
+/** Decode a projection Opus packet.
+  * @param st <tt>OpusProjectionDecoder*</tt>: Projection decoder state.
+  * @param[in] data <tt>const unsigned char*</tt>: Input payload.
+  *                                                Use a <code>NULL</code>
+  *                                                pointer to indicate packet
+  *                                                loss.
+  * @param len <tt>opus_int32</tt>: Number of bytes in payload.
+  * @param[out] pcm <tt>opus_int32*</tt>: Output signal, with interleaved
+  *                                       samples representing (or slightly exceeding) 24-bit values.
+  *                                       This must contain room for
+  *                                       <code>frame_size*channels</code>
+  *                                       samples.
+  * @param frame_size <tt>int</tt>: The number of samples per channel of
+  *                                 available space in \a pcm.
+  *                                 If this is less than the maximum packet duration
+  *                                 (120 ms; 5760 for 48kHz), this function will not be capable
+  *                                 of decoding some packets. In the case of PLC (data==NULL)
+  *                                 or FEC (decode_fec=1), then frame_size needs to be exactly
+  *                                 the duration of audio that is missing, otherwise the
+  *                                 decoder will not be in the optimal state to decode the
+  *                                 next incoming packet. For the PLC and FEC cases, frame_size
+  *                                 <b>must</b> be a multiple of 2.5 ms.
+  * @param decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band
+  *                                 forward error correction data be decoded.
+  *                                 If no such data is available, the frame is
+  *                                 decoded as if it were lost.
+  * @returns Number of samples decoded on success or a negative error code
+  *          (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_projection_decode24(
+    OpusProjectionDecoder *st,
+    const unsigned char *data,
+    opus_int32 len,
+    opus_int32 *pcm,
+    int frame_size,
+    int decode_fec
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
 
 /** Decode a projection Opus packet with floating point output.
   * @param st <tt>OpusProjectionDecoder*</tt>: Projection decoder state.