mirror of https://github.com/thesofproject/sof.git
copier-dai: refactor copier parameters
Move dai parameter related code to copier_dai.c with new functions. Signed-off-by: Baofeng Tian <baofeng.tian@intel.com>
This commit is contained in:
Baofeng Tian
Kai Vehmanen
parent
ee318548d7
commit
97f78ea3f1
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@ -392,88 +392,6 @@ pcm_converter_func get_converter_func(const struct ipc4_audio_format *in_fmt,
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return pcm_get_conversion_vc_function(in, in_valid, out, out_valid, type, dir);
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}
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static int copy_single_channel_c16(const struct audio_stream __sparse_cache *src,
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unsigned int src_channel,
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struct audio_stream __sparse_cache *dst,
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unsigned int dst_channel, unsigned int frame_count)
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{
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int16_t *r_ptr = (int16_t *)audio_stream_get_rptr(src) + src_channel;
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int16_t *w_ptr = (int16_t *)audio_stream_get_wptr(dst) + dst_channel;
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/* We have to iterate over frames here. However, tracking frames requires using
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* of expensive division operations (e.g., inside audio_stream_frames_without_wrap()).
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* So let's track samples instead. Since we only copy one channel, src_stream_sample_count
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* is NOT number of samples we need to copy but total samples for all channels. We just
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* track them to know when to stop.
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*/
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int src_stream_sample_count = frame_count * audio_stream_get_channels(src);
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while (src_stream_sample_count) {
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int src_samples_without_wrap;
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int16_t *r_end_ptr, *r_ptr_before_loop;
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r_ptr = audio_stream_wrap(src, r_ptr);
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w_ptr = audio_stream_wrap(dst, w_ptr);
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src_samples_without_wrap = audio_stream_samples_without_wrap_s16(src, r_ptr);
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r_end_ptr = src_stream_sample_count < src_samples_without_wrap ?
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r_ptr + src_stream_sample_count : (int16_t *)audio_stream_get_end_addr(src);
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r_ptr_before_loop = r_ptr;
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do {
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*w_ptr = *r_ptr;
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r_ptr += audio_stream_get_channels(src);
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w_ptr += audio_stream_get_channels(dst);
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} while (r_ptr < r_end_ptr && w_ptr < (int16_t *)audio_stream_get_end_addr(dst));
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src_stream_sample_count -= r_ptr - r_ptr_before_loop;
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}
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return 0;
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}
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static int copy_single_channel_c32(const struct audio_stream __sparse_cache *src,
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unsigned int src_channel,
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struct audio_stream __sparse_cache *dst,
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unsigned int dst_channel, unsigned int frame_count)
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{
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int32_t *r_ptr = (int32_t *)audio_stream_get_rptr(src) + src_channel;
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int32_t *w_ptr = (int32_t *)audio_stream_get_wptr(dst) + dst_channel;
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/* We have to iterate over frames here. However, tracking frames requires using
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* of expensive division operations (e.g., inside audio_stream_frames_without_wrap()).
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* So let's track samples instead. Since we only copy one channel, src_stream_sample_count
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* is NOT number of samples we need to copy but total samples for all channels. We just
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* track them to know when to stop.
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*/
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int src_stream_sample_count = frame_count * audio_stream_get_channels(src);
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while (src_stream_sample_count) {
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int src_samples_without_wrap;
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int32_t *r_end_ptr, *r_ptr_before_loop;
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r_ptr = audio_stream_wrap(src, r_ptr);
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w_ptr = audio_stream_wrap(dst, w_ptr);
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src_samples_without_wrap = audio_stream_samples_without_wrap_s32(src, r_ptr);
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r_end_ptr = src_stream_sample_count < src_samples_without_wrap ?
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r_ptr + src_stream_sample_count : (int32_t *)audio_stream_get_end_addr(src);
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r_ptr_before_loop = r_ptr;
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do {
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*w_ptr = *r_ptr;
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r_ptr += audio_stream_get_channels(src);
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w_ptr += audio_stream_get_channels(dst);
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} while (r_ptr < r_end_ptr && w_ptr < (int32_t *)audio_stream_get_end_addr(dst));
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src_stream_sample_count -= r_ptr - r_ptr_before_loop;
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}
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return 0;
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}
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static int copier_prepare(struct comp_dev *dev)
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{
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struct copier_data *cd = comp_get_drvdata(dev);
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@ -869,105 +787,15 @@ static int copier_copy(struct comp_dev *dev)
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return ret;
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}
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static void update_buffer_format(struct comp_buffer __sparse_cache *buf_c,
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const struct ipc4_audio_format *fmt)
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{
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enum sof_ipc_frame valid_fmt, frame_fmt;
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int i;
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buf_c->stream.channels = fmt->channels_count;
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buf_c->stream.rate = fmt->sampling_frequency;
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audio_stream_fmt_conversion(fmt->depth,
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fmt->valid_bit_depth,
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&frame_fmt, &valid_fmt,
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fmt->s_type);
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buf_c->stream.frame_fmt = frame_fmt;
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buf_c->stream.valid_sample_fmt = valid_fmt;
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buf_c->buffer_fmt = fmt->interleaving_style;
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for (i = 0; i < SOF_IPC_MAX_CHANNELS; i++)
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buf_c->chmap[i] = (fmt->ch_map >> i * 4) & 0xf;
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buf_c->hw_params_configured = true;
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}
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/* configure the DMA params */
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static int copier_params(struct comp_dev *dev, struct sof_ipc_stream_params *params)
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{
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struct copier_data *cd = comp_get_drvdata(dev);
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const struct ipc4_audio_format *in_fmt = &cd->config.base.audio_fmt;
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const struct ipc4_audio_format *out_fmt = &cd->config.out_fmt;
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struct comp_buffer *sink, *source;
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struct comp_buffer __sparse_cache *sink_c, *source_c;
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struct list_item *sink_list;
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enum sof_ipc_frame in_bits, in_valid_bits, out_bits, out_valid_bits;
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int i, ret = 0;
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comp_dbg(dev, "copier_params()");
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memset(params, 0, sizeof(*params));
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params->direction = cd->direction;
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params->channels = cd->config.base.audio_fmt.channels_count;
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params->rate = cd->config.base.audio_fmt.sampling_frequency;
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params->sample_container_bytes = cd->config.base.audio_fmt.depth / 8;
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params->sample_valid_bytes = cd->config.base.audio_fmt.valid_bit_depth / 8;
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params->stream_tag = cd->config.gtw_cfg.node_id.f.v_index + 1;
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params->frame_fmt = dev->ipc_config.frame_fmt;
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params->buffer_fmt = cd->config.base.audio_fmt.interleaving_style;
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params->buffer.size = cd->config.base.ibs;
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/* disable ipc3 stream position */
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params->no_stream_position = 1;
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/* update each sink format */
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list_for_item(sink_list, &dev->bsink_list) {
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int j;
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sink = container_of(sink_list, struct comp_buffer, source_list);
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sink_c = buffer_acquire(sink);
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j = IPC4_SINK_QUEUE_ID(sink_c->id);
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update_buffer_format(sink_c, &cd->out_fmt[j]);
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buffer_release(sink_c);
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}
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/*
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* force update the source buffer format to cover cases where the source module
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* fails to set the sink buffer params
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*/
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if (!list_is_empty(&dev->bsource_list)) {
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struct ipc4_audio_format *in_fmt;
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source = list_first_item(&dev->bsource_list, struct comp_buffer, sink_list);
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source_c = buffer_acquire(source);
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in_fmt = &cd->config.base.audio_fmt;
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update_buffer_format(source_c, in_fmt);
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buffer_release(source_c);
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}
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/* update params for the DMA buffer */
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switch (dev->ipc_config.type) {
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case SOF_COMP_HOST:
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if (cd->ipc_gtw || params->direction == SOF_IPC_STREAM_PLAYBACK)
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break;
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COMPILER_FALLTHROUGH;
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case SOF_COMP_DAI:
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if (dev->ipc_config.type == SOF_COMP_DAI &&
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(cd->endpoint_num > 1 || params->direction == SOF_IPC_STREAM_CAPTURE))
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break;
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params->buffer.size = cd->config.base.obs;
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params->sample_container_bytes = cd->out_fmt->depth / 8;
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params->sample_valid_bytes = cd->out_fmt->valid_bit_depth / 8;
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break;
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default:
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break;
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}
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copier_update_params(cd, dev, params);
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for (i = 0; i < cd->endpoint_num; i++) {
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switch (dev->ipc_config.type) {
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@ -979,70 +807,8 @@ static int copier_params(struct comp_dev *dev, struct sof_ipc_stream_params *par
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ret = copier_ipcgtw_params(cd->ipcgtw_data, dev, params);
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break;
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case SOF_COMP_DAI:
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{
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struct comp_buffer __sparse_cache *buf_c;
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struct sof_ipc_stream_params demuxed_params = *params;
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int container_size;
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int j;
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if (cd->endpoint_num == 1) {
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ret = dai_zephyr_params(cd->dd[0], dev, params);
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/*
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* dai_zephyr_params assigns the conversion function
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* based on the input/output formats but does not take
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* the valid bits into account. So change the conversion
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* function if the valid bits are different from the
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* container size.
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*/
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audio_stream_fmt_conversion(in_fmt->depth,
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in_fmt->valid_bit_depth,
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&in_bits, &in_valid_bits,
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in_fmt->s_type);
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audio_stream_fmt_conversion(out_fmt->depth,
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out_fmt->valid_bit_depth,
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&out_bits, &out_valid_bits,
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out_fmt->s_type);
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if (in_bits != in_valid_bits || out_bits != out_valid_bits)
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cd->dd[0]->process =
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cd->converter[IPC4_COPIER_GATEWAY_PIN];
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ret = copier_dai_params(cd, dev, params, i);
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break;
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}
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/* For ALH multi-gateway case, params->channels is a total multiplexed
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* number of channels. Demultiplexed number of channels for each individual
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* gateway comes in blob's struct ipc4_alh_multi_gtw_cfg.
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*/
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demuxed_params.channels = cd->channels[i];
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ret = dai_zephyr_params(cd->dd[i], dev, &demuxed_params);
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if (ret < 0)
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return ret;
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buf_c = buffer_acquire(cd->dd[i]->dma_buffer);
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for (j = 0; j < SOF_IPC_MAX_CHANNELS; j++)
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buf_c->chmap[j] = (cd->chan_map[i] >> j * 4) & 0xf;
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buffer_release(buf_c);
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/* set channel copy func */
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buf_c = buffer_acquire(cd->multi_endpoint_buffer);
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container_size = audio_stream_sample_bytes(&buf_c->stream);
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buffer_release(buf_c);
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switch (container_size) {
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case 2:
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cd->dd[i]->process = copy_single_channel_c16;
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break;
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case 4:
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cd->dd[i]->process = copy_single_channel_c32;
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break;
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default:
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comp_err(dev, "Unexpected container size: %d", container_size);
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return -EINVAL;
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}
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break;
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}
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default:
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break;
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}
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@ -303,3 +303,156 @@ int copier_dai_prepare(struct comp_dev *dev, struct copier_data *cd)
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return 0;
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}
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static int copy_single_channel_c16(const struct audio_stream __sparse_cache *src,
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unsigned int src_channel,
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struct audio_stream __sparse_cache *dst,
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unsigned int dst_channel, unsigned int frame_count)
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{
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int16_t *r_ptr = (int16_t *)audio_stream_get_rptr(src) + src_channel;
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int16_t *w_ptr = (int16_t *)audio_stream_get_wptr(dst) + dst_channel;
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/* We have to iterate over frames here. However, tracking frames requires using
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* of expensive division operations (e.g., inside audio_stream_frames_without_wrap()).
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* So let's track samples instead. Since we only copy one channel, src_stream_sample_count
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* is NOT number of samples we need to copy but total samples for all channels. We just
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* track them to know when to stop.
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*/
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int src_stream_sample_count = frame_count * audio_stream_get_channels(src);
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while (src_stream_sample_count) {
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int src_samples_without_wrap;
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int16_t *r_end_ptr, *r_ptr_before_loop;
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r_ptr = audio_stream_wrap(src, r_ptr);
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w_ptr = audio_stream_wrap(dst, w_ptr);
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src_samples_without_wrap = audio_stream_samples_without_wrap_s16(src, r_ptr);
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r_end_ptr = src_stream_sample_count < src_samples_without_wrap ?
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r_ptr + src_stream_sample_count : (int16_t *)audio_stream_get_end_addr(src);
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r_ptr_before_loop = r_ptr;
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do {
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*w_ptr = *r_ptr;
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r_ptr += audio_stream_get_channels(src);
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w_ptr += audio_stream_get_channels(dst);
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} while (r_ptr < r_end_ptr && w_ptr < (int16_t *)audio_stream_get_end_addr(dst));
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src_stream_sample_count -= r_ptr - r_ptr_before_loop;
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}
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return 0;
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}
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static int copy_single_channel_c32(const struct audio_stream __sparse_cache *src,
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unsigned int src_channel,
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struct audio_stream __sparse_cache *dst,
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unsigned int dst_channel, unsigned int frame_count)
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{
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int32_t *r_ptr = (int32_t *)audio_stream_get_rptr(src) + src_channel;
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int32_t *w_ptr = (int32_t *)audio_stream_get_wptr(dst) + dst_channel;
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/* We have to iterate over frames here. However, tracking frames requires using
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* of expensive division operations (e.g., inside audio_stream_frames_without_wrap()).
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* So let's track samples instead. Since we only copy one channel, src_stream_sample_count
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* is NOT number of samples we need to copy but total samples for all channels. We just
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* track them to know when to stop.
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*/
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int src_stream_sample_count = frame_count * audio_stream_get_channels(src);
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while (src_stream_sample_count) {
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int src_samples_without_wrap;
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int32_t *r_end_ptr, *r_ptr_before_loop;
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r_ptr = audio_stream_wrap(src, r_ptr);
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w_ptr = audio_stream_wrap(dst, w_ptr);
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src_samples_without_wrap = audio_stream_samples_without_wrap_s32(src, r_ptr);
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r_end_ptr = src_stream_sample_count < src_samples_without_wrap ?
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r_ptr + src_stream_sample_count : (int32_t *)audio_stream_get_end_addr(src);
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r_ptr_before_loop = r_ptr;
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do {
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*w_ptr = *r_ptr;
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r_ptr += audio_stream_get_channels(src);
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w_ptr += audio_stream_get_channels(dst);
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} while (r_ptr < r_end_ptr && w_ptr < (int32_t *)audio_stream_get_end_addr(dst));
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src_stream_sample_count -= r_ptr - r_ptr_before_loop;
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}
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return 0;
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}
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int copier_dai_params(struct copier_data *cd, struct comp_dev *dev,
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struct sof_ipc_stream_params *params, int dai_index)
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{
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struct comp_buffer __sparse_cache *buf_c;
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struct sof_ipc_stream_params demuxed_params = *params;
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const struct ipc4_audio_format *in_fmt = &cd->config.base.audio_fmt;
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const struct ipc4_audio_format *out_fmt = &cd->config.out_fmt;
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enum sof_ipc_frame in_bits, in_valid_bits, out_bits, out_valid_bits;
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int container_size;
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int j, ret;
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if (cd->endpoint_num == 1) {
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ret = dai_zephyr_params(cd->dd[0], dev, params);
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/*
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* dai_zephyr_params assigns the conversion function
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* based on the input/output formats but does not take
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* the valid bits into account. So change the conversion
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* function if the valid bits are different from the
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* container size.
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*/
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audio_stream_fmt_conversion(in_fmt->depth,
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in_fmt->valid_bit_depth,
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&in_bits, &in_valid_bits,
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in_fmt->s_type);
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audio_stream_fmt_conversion(out_fmt->depth,
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out_fmt->valid_bit_depth,
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&out_bits, &out_valid_bits,
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out_fmt->s_type);
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if (in_bits != in_valid_bits || out_bits != out_valid_bits)
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cd->dd[0]->process =
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cd->converter[IPC4_COPIER_GATEWAY_PIN];
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return ret;
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}
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/* For ALH multi-gateway case, params->channels is a total multiplexed
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* number of channels. Demultiplexed number of channels for each individual
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* gateway comes in blob's struct ipc4_alh_multi_gtw_cfg.
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*/
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demuxed_params.channels = cd->channels[dai_index];
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ret = dai_zephyr_params(cd->dd[dai_index], dev, &demuxed_params);
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if (ret < 0)
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return ret;
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buf_c = buffer_acquire(cd->dd[dai_index]->dma_buffer);
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for (j = 0; j < SOF_IPC_MAX_CHANNELS; j++)
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buf_c->chmap[j] = (cd->chan_map[dai_index] >> j * 4) & 0xf;
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buffer_release(buf_c);
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/* set channel copy func */
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buf_c = buffer_acquire(cd->multi_endpoint_buffer);
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container_size = audio_stream_sample_bytes(&buf_c->stream);
|
||||
buffer_release(buf_c);
|
||||
|
||||
switch (container_size) {
|
||||
case 2:
|
||||
cd->dd[dai_index]->process = copy_single_channel_c16;
|
||||
break;
|
||||
case 4:
|
||||
cd->dd[dai_index]->process = copy_single_channel_c32;
|
||||
break;
|
||||
default:
|
||||
comp_err(dev, "Unexpected container size: %d", container_size);
|
||||
return -EINVAL;
|
||||
}
|
||||
|
||||
return ret;
|
||||
}
|
||||
|
|
|
|||
|
|
@ -5,11 +5,11 @@
|
|||
// Author: Andrula Song <xiaoyuan.song@intel.com>
|
||||
|
||||
#include <ipc4/copier.h>
|
||||
#include <sof/audio/component_ext.h>
|
||||
|
||||
#ifdef COPIER_GENERIC
|
||||
|
||||
#include <sof/audio/buffer.h>
|
||||
#include <sof/audio/component_ext.h>
|
||||
#include <sof/audio/format.h>
|
||||
#include <sof/audio/pipeline.h>
|
||||
#include <sof/audio/component.h>
|
||||
|
|
@ -56,3 +56,98 @@ int apply_attenuation(struct comp_dev *dev, struct copier_data *cd,
|
|||
}
|
||||
|
||||
#endif
|
||||
|
||||
void update_buffer_format(struct comp_buffer __sparse_cache *buf_c,
|
||||
const struct ipc4_audio_format *fmt)
|
||||
{
|
||||
enum sof_ipc_frame valid_fmt, frame_fmt;
|
||||
int i;
|
||||
|
||||
buf_c->stream.channels = fmt->channels_count;
|
||||
buf_c->stream.rate = fmt->sampling_frequency;
|
||||
audio_stream_fmt_conversion(fmt->depth,
|
||||
fmt->valid_bit_depth,
|
||||
&frame_fmt, &valid_fmt,
|
||||
fmt->s_type);
|
||||
|
||||
buf_c->stream.frame_fmt = frame_fmt;
|
||||
buf_c->stream.valid_sample_fmt = valid_fmt;
|
||||
|
||||
buf_c->buffer_fmt = fmt->interleaving_style;
|
||||
|
||||
for (i = 0; i < SOF_IPC_MAX_CHANNELS; i++)
|
||||
buf_c->chmap[i] = (fmt->ch_map >> i * 4) & 0xf;
|
||||
|
||||
buf_c->hw_params_configured = true;
|
||||
}
|
||||
|
||||
void copier_update_params(struct copier_data *cd, struct comp_dev *dev,
|
||||
struct sof_ipc_stream_params *params)
|
||||
{
|
||||
struct comp_buffer *sink, *source;
|
||||
struct comp_buffer __sparse_cache *sink_c, *source_c;
|
||||
struct list_item *sink_list;
|
||||
|
||||
memset(params, 0, sizeof(*params));
|
||||
params->direction = cd->direction;
|
||||
params->channels = cd->config.base.audio_fmt.channels_count;
|
||||
params->rate = cd->config.base.audio_fmt.sampling_frequency;
|
||||
params->sample_container_bytes = cd->config.base.audio_fmt.depth / 8;
|
||||
params->sample_valid_bytes = cd->config.base.audio_fmt.valid_bit_depth / 8;
|
||||
|
||||
params->stream_tag = cd->config.gtw_cfg.node_id.f.v_index + 1;
|
||||
params->frame_fmt = dev->ipc_config.frame_fmt;
|
||||
params->buffer_fmt = cd->config.base.audio_fmt.interleaving_style;
|
||||
params->buffer.size = cd->config.base.ibs;
|
||||
|
||||
/* disable ipc3 stream position */
|
||||
params->no_stream_position = 1;
|
||||
|
||||
/* update each sink format */
|
||||
list_for_item(sink_list, &dev->bsink_list) {
|
||||
int j;
|
||||
|
||||
sink = container_of(sink_list, struct comp_buffer, source_list);
|
||||
sink_c = buffer_acquire(sink);
|
||||
|
||||
j = IPC4_SINK_QUEUE_ID(sink_c->id);
|
||||
|
||||
update_buffer_format(sink_c, &cd->out_fmt[j]);
|
||||
|
||||
buffer_release(sink_c);
|
||||
}
|
||||
|
||||
/*
|
||||
* force update the source buffer format to cover cases where the source module
|
||||
* fails to set the sink buffer params
|
||||
*/
|
||||
if (!list_is_empty(&dev->bsource_list)) {
|
||||
struct ipc4_audio_format *in_fmt;
|
||||
|
||||
source = list_first_item(&dev->bsource_list, struct comp_buffer, sink_list);
|
||||
source_c = buffer_acquire(source);
|
||||
|
||||
in_fmt = &cd->config.base.audio_fmt;
|
||||
update_buffer_format(source_c, in_fmt);
|
||||
|
||||
buffer_release(source_c);
|
||||
}
|
||||
|
||||
/* update params for the DMA buffer */
|
||||
switch (dev->ipc_config.type) {
|
||||
case SOF_COMP_HOST:
|
||||
if (cd->ipc_gtw || params->direction == SOF_IPC_STREAM_PLAYBACK)
|
||||
break;
|
||||
COMPILER_FALLTHROUGH;
|
||||
case SOF_COMP_DAI:
|
||||
if (dev->ipc_config.type == SOF_COMP_DAI &&
|
||||
(cd->endpoint_num > 1 || params->direction == SOF_IPC_STREAM_CAPTURE))
|
||||
break;
|
||||
params->buffer.size = cd->config.base.obs;
|
||||
params->sample_container_bytes = cd->out_fmt->depth / 8;
|
||||
params->sample_valid_bytes = cd->out_fmt->valid_bit_depth / 8;
|
||||
break;
|
||||
default:
|
||||
break;
|
||||
}
|
||||
}
|
||||
|
|
|
|||
|
|
@ -283,4 +283,9 @@ int create_endpoint_buffer(struct comp_dev *parent_dev,
|
|||
enum sof_ipc_stream_direction
|
||||
get_gateway_direction(enum ipc4_connector_node_id_type node_id_type);
|
||||
|
||||
void update_buffer_format(struct comp_buffer __sparse_cache *buf_c,
|
||||
const struct ipc4_audio_format *fmt);
|
||||
|
||||
void copier_update_params(struct copier_data *cd, struct comp_dev *dev,
|
||||
struct sof_ipc_stream_params *params);
|
||||
#endif
|
||||
|
|
|
|||
|
|
@ -65,4 +65,7 @@ int copier_dai_create(struct comp_dev *parent_dev, struct copier_data *cd,
|
|||
void copier_dai_free(struct copier_data *cd);
|
||||
|
||||
int copier_dai_prepare(struct comp_dev *dev, struct copier_data *cd);
|
||||
|
||||
int copier_dai_params(struct copier_data *cd, struct comp_dev *dev,
|
||||
struct sof_ipc_stream_params *params, int dai_index);
|
||||
#endif /* __SOF_LIB_DAI_COPIER_H__ */
|
||||
|
|
|
|||
Loading…
Reference in New Issue