From feb8eb935bd69ce855533bf9a4316e3025602c1a Mon Sep 17 00:00:00 2001 From: Jeff Epler Date: Wed, 4 Dec 2019 09:31:52 -0600 Subject: audiosample: convert to use a protocol This eases addition of new sample sources, since the manual virtual function dispatch functions are just calls via a protocol --- shared-module/audiocore/RawSample.c | 6 +++ shared-module/audiocore/__init__.c | 92 ++++++------------------------------- shared-module/audiocore/__init__.h | 35 +++++++++++--- shared-module/audiomixer/Mixer.c | 8 ++++ 4 files changed, 56 insertions(+), 85 deletions(-) (limited to 'shared-module') diff --git a/shared-module/audiocore/RawSample.c b/shared-module/audiocore/RawSample.c index d6bf3a567..a69ddeb65 100644 --- a/shared-module/audiocore/RawSample.c +++ b/shared-module/audiocore/RawSample.c @@ -60,6 +60,12 @@ void common_hal_audioio_rawsample_set_sample_rate(audioio_rawsample_obj_t* self, uint32_t sample_rate) { self->sample_rate = sample_rate; } +uint8_t common_hal_audioio_rawsample_get_bits_per_sample(audioio_rawsample_obj_t* self) { + return self->bits_per_sample; +} +uint8_t common_hal_audioio_rawsample_get_channel_count(audioio_rawsample_obj_t* self) { + return self->channel_count; +} void audioio_rawsample_reset_buffer(audioio_rawsample_obj_t* self, bool single_channel, diff --git a/shared-module/audiocore/__init__.c b/shared-module/audiocore/__init__.c index f9762676f..1553dbe96 100644 --- a/shared-module/audiocore/__init__.c +++ b/shared-module/audiocore/__init__.c @@ -36,103 +36,37 @@ #include "shared-module/audiomixer/Mixer.h" uint32_t audiosample_sample_rate(mp_obj_t sample_obj) { - if (MP_OBJ_IS_TYPE(sample_obj, &audioio_rawsample_type)) { - audioio_rawsample_obj_t* sample = MP_OBJ_TO_PTR(sample_obj); - return sample->sample_rate; - } else if (MP_OBJ_IS_TYPE(sample_obj, &audioio_wavefile_type)) { - audioio_wavefile_obj_t* file = MP_OBJ_TO_PTR(sample_obj); - return file->sample_rate; - #if CIRCUITPY_AUDIOMIXER - } else if (MP_OBJ_IS_TYPE(sample_obj, &audiomixer_mixer_type)) { - audiomixer_mixer_obj_t* mixer = MP_OBJ_TO_PTR(sample_obj); - return mixer->sample_rate; - #endif - } - return 16000; + const audiosample_p_t *proto = mp_proto_get_or_throw(MP_QSTR_protocol_audiosample, sample_obj); + return proto->sample_rate(MP_OBJ_TO_PTR(sample_obj)); } uint8_t audiosample_bits_per_sample(mp_obj_t sample_obj) { - if (MP_OBJ_IS_TYPE(sample_obj, &audioio_rawsample_type)) { - audioio_rawsample_obj_t* sample = MP_OBJ_TO_PTR(sample_obj); - return sample->bits_per_sample; - } else if (MP_OBJ_IS_TYPE(sample_obj, &audioio_wavefile_type)) { - audioio_wavefile_obj_t* file = MP_OBJ_TO_PTR(sample_obj); - return file->bits_per_sample; - #if CIRCUITPY_AUDIOMIXER - } else if (MP_OBJ_IS_TYPE(sample_obj, &audiomixer_mixer_type)) { - audiomixer_mixer_obj_t* mixer = MP_OBJ_TO_PTR(sample_obj); - return mixer->bits_per_sample; - #endif - } - return 8; + const audiosample_p_t *proto = mp_proto_get_or_throw(MP_QSTR_protocol_audiosample, sample_obj); + return proto->bits_per_sample(MP_OBJ_TO_PTR(sample_obj)); } uint8_t audiosample_channel_count(mp_obj_t sample_obj) { - if (MP_OBJ_IS_TYPE(sample_obj, &audioio_rawsample_type)) { - audioio_rawsample_obj_t* sample = MP_OBJ_TO_PTR(sample_obj); - return sample->channel_count; - } else if (MP_OBJ_IS_TYPE(sample_obj, &audioio_wavefile_type)) { - audioio_wavefile_obj_t* file = MP_OBJ_TO_PTR(sample_obj); - return file->channel_count; - #if CIRCUITPY_AUDIOMIXER - } else if (MP_OBJ_IS_TYPE(sample_obj, &audiomixer_mixer_type)) { - audiomixer_mixer_obj_t* mixer = MP_OBJ_TO_PTR(sample_obj); - return mixer->channel_count; - #endif - } - return 1; + const audiosample_p_t *proto = mp_proto_get_or_throw(MP_QSTR_protocol_audiosample, sample_obj); + return proto->channel_count(MP_OBJ_TO_PTR(sample_obj)); } void audiosample_reset_buffer(mp_obj_t sample_obj, bool single_channel, uint8_t audio_channel) { - if (MP_OBJ_IS_TYPE(sample_obj, &audioio_rawsample_type)) { - audioio_rawsample_obj_t* sample = MP_OBJ_TO_PTR(sample_obj); - audioio_rawsample_reset_buffer(sample, single_channel, audio_channel); - } else if (MP_OBJ_IS_TYPE(sample_obj, &audioio_wavefile_type)) { - audioio_wavefile_obj_t* file = MP_OBJ_TO_PTR(sample_obj); - audioio_wavefile_reset_buffer(file, single_channel, audio_channel); - #if CIRCUITPY_AUDIOMIXER - } else if (MP_OBJ_IS_TYPE(sample_obj, &audiomixer_mixer_type)) { - audiomixer_mixer_obj_t* file = MP_OBJ_TO_PTR(sample_obj); - audiomixer_mixer_reset_buffer(file, single_channel, audio_channel); - #endif - } + const audiosample_p_t *proto = mp_proto_get_or_throw(MP_QSTR_protocol_audiosample, sample_obj); + proto->reset_buffer(MP_OBJ_TO_PTR(sample_obj)); } audioio_get_buffer_result_t audiosample_get_buffer(mp_obj_t sample_obj, bool single_channel, uint8_t channel, uint8_t** buffer, uint32_t* buffer_length) { - if (MP_OBJ_IS_TYPE(sample_obj, &audioio_rawsample_type)) { - audioio_rawsample_obj_t* sample = MP_OBJ_TO_PTR(sample_obj); - return audioio_rawsample_get_buffer(sample, single_channel, channel, buffer, buffer_length); - } else if (MP_OBJ_IS_TYPE(sample_obj, &audioio_wavefile_type)) { - audioio_wavefile_obj_t* file = MP_OBJ_TO_PTR(sample_obj); - return audioio_wavefile_get_buffer(file, single_channel, channel, buffer, buffer_length); - #if CIRCUITPY_AUDIOMIXER - } else if (MP_OBJ_IS_TYPE(sample_obj, &audiomixer_mixer_type)) { - audiomixer_mixer_obj_t* file = MP_OBJ_TO_PTR(sample_obj); - return audiomixer_mixer_get_buffer(file, single_channel, channel, buffer, buffer_length); - #endif - } - return GET_BUFFER_DONE; + const audiosample_p_t *proto = mp_proto_get_or_throw(MP_QSTR_protocol_audiosample, sample_obj); + return proto->get_buffer(MP_OBJ_TO_PTR(sample_obj), single_channel, channel, buffer, buffer_length); } void audiosample_get_buffer_structure(mp_obj_t sample_obj, bool single_channel, bool* single_buffer, bool* samples_signed, uint32_t* max_buffer_length, uint8_t* spacing) { - if (MP_OBJ_IS_TYPE(sample_obj, &audioio_rawsample_type)) { - audioio_rawsample_obj_t* sample = MP_OBJ_TO_PTR(sample_obj); - audioio_rawsample_get_buffer_structure(sample, single_channel, single_buffer, - samples_signed, max_buffer_length, spacing); - } else if (MP_OBJ_IS_TYPE(sample_obj, &audioio_wavefile_type)) { - audioio_wavefile_obj_t* file = MP_OBJ_TO_PTR(sample_obj); - audioio_wavefile_get_buffer_structure(file, single_channel, single_buffer, samples_signed, - max_buffer_length, spacing); - #if CIRCUITPY_AUDIOMIXER - } else if (MP_OBJ_IS_TYPE(sample_obj, &audiomixer_mixer_type)) { - audiomixer_mixer_obj_t* file = MP_OBJ_TO_PTR(sample_obj); - audiomixer_mixer_get_buffer_structure(file, single_channel, single_buffer, samples_signed, - max_buffer_length, spacing); - #endif - } + const audiosample_p_t *proto = mp_proto_get_or_throw(MP_QSTR_protocol_audiosample, sample_obj); + proto->get_buffer_structure(MP_OBJ_TO_PTR(sample_obj), single_channel, single_buffer, + samples_signed, max_buffer_length, spacing); } diff --git a/shared-module/audiocore/__init__.h b/shared-module/audiocore/__init__.h index 7aa0c1824..4b306f8ba 100644 --- a/shared-module/audiocore/__init__.h +++ b/shared-module/audiocore/__init__.h @@ -31,6 +31,7 @@ #include #include "py/obj.h" +#include "py/proto.h" typedef enum { GET_BUFFER_DONE, // No more data to read @@ -38,15 +39,37 @@ typedef enum { GET_BUFFER_ERROR, // Error while reading data. } audioio_get_buffer_result_t; -uint32_t audiosample_sample_rate(mp_obj_t sample_obj); -uint8_t audiosample_bits_per_sample(mp_obj_t sample_obj); -uint8_t audiosample_channel_count(mp_obj_t sample_obj); -void audiosample_reset_buffer(mp_obj_t sample_obj, bool single_channel, uint8_t audio_channel); -audioio_get_buffer_result_t audiosample_get_buffer(mp_obj_t sample_obj, +typedef uint32_t (*audiosample_sample_rate_fun)(void* sample_obj); +typedef uint8_t (*audiosample_bits_per_sample_fun)(void* sample_obj); +typedef uint8_t (*audiosample_channel_count_fun)(void* sample_obj); +typedef void (*audiosample_reset_buffer_fun)(void* sample_obj); +typedef audioio_get_buffer_result_t (*audiosample_get_buffer_fun)(void* sample_obj, + bool single_channel, uint8_t channel, uint8_t** buffer, + uint32_t* buffer_length); +typedef void (*audiosample_get_buffer_structure_fun)(void* sample_obj, + bool single_channel, bool* single_buffer, + bool* samples_signed, uint32_t *max_buffer_length, + uint8_t* spacing); + +typedef struct _audiosample_p_t { + MP_PROTOCOL_HEAD // MP_QSTR_protocol_audiosample + audiosample_sample_rate_fun sample_rate; + audiosample_bits_per_sample_fun bits_per_sample; + audiosample_channel_count_fun channel_count; + audiosample_reset_buffer_fun reset_buffer; + audiosample_get_buffer_fun get_buffer; + audiosample_get_buffer_structure_fun get_buffer_structure; +} audiosample_p_t; + +uint32_t audiosample_sample_rate(void* sample_obj); +uint8_t audiosample_bits_per_sample(void* sample_obj); +uint8_t audiosample_channel_count(void* sample_obj); +void audiosample_reset_buffer(void* sample_obj, bool single_channel, uint8_t audio_channel); +audioio_get_buffer_result_t audiosample_get_buffer(void* sample_obj, bool single_channel, uint8_t channel, uint8_t** buffer, uint32_t* buffer_length); -void audiosample_get_buffer_structure(mp_obj_t sample_obj, bool single_channel, +void audiosample_get_buffer_structure(void* sample_obj, bool single_channel, bool* single_buffer, bool* samples_signed, uint32_t* max_buffer_length, uint8_t* spacing); diff --git a/shared-module/audiomixer/Mixer.c b/shared-module/audiomixer/Mixer.c index a52489f85..afa3a0632 100644 --- a/shared-module/audiomixer/Mixer.c +++ b/shared-module/audiomixer/Mixer.c @@ -76,6 +76,14 @@ uint32_t common_hal_audiomixer_mixer_get_sample_rate(audiomixer_mixer_obj_t* sel return self->sample_rate; } +uint8_t common_hal_audiomixer_mixer_get_channel_count(audiomixer_mixer_obj_t* self) { + return self->channel_count; +} + +uint8_t common_hal_audiomixer_mixer_get_bits_per_sample(audiomixer_mixer_obj_t* self) { + return self->bits_per_sample; +} + bool common_hal_audiomixer_mixer_get_playing(audiomixer_mixer_obj_t* self) { for (uint8_t v = 0; v < self->voice_count; v++) { if (common_hal_audiomixer_mixervoice_get_playing(MP_OBJ_TO_PTR(self->voice[v]))) { -- cgit v1.2.3