Hi Andrei,
On Thu, May 8, 2014 at 11:01 AM, Andrei Emeltchenko
<Andrei.Emeltchenko.news@xxxxxxxxx> wrote:
> From: Andrei Emeltchenko <andrei.emeltchenko@xxxxxxxxx>
>
> Function adds audio_open_output_stream() and sets dummy callbacks.
> ---
> android/hal-sco.c | 179 +++++++++++++++++++++++++++++++++++++++++++++++++++++-
> 1 file changed, 176 insertions(+), 3 deletions(-)
>
> diff --git a/android/hal-sco.c b/android/hal-sco.c
> index c7ba504..bc7d987 100644
> --- a/android/hal-sco.c
> +++ b/android/hal-sco.c
> @@ -25,22 +25,195 @@
>
> #include "hal-log.h"
>
> +#define AUDIO_STREAM_DEFAULT_RATE 44100
> +#define AUDIO_STREAM_DEFAULT_FORMAT AUDIO_FORMAT_PCM_16_BIT
> +
> +#define OUT_BUFFER_SIZE 2560
> +
> +struct hsp_audio_config {
> + uint32_t rate;
> + uint32_t channels;
> + audio_format_t format;
> +};
> +
> +struct hsp_stream_out {
> + struct audio_stream_out stream;
> + struct hsp_audio_config cfg;
> +};
> +
> struct hsp_audio_dev {
> struct audio_hw_device dev;
> - struct a2dp_stream_out *out;
> + struct hsp_stream_out *out;
> };
Please rename hsp to sco, also fix where you actually introduced
a2dp_stream_out.
Tip: you can use git rebase -i origin/master change the commit that
want to fix from pick to edit once you are done do git add and git
rebase --continue
> +/* Audio stream functions */
> +
> +static ssize_t out_write(struct audio_stream_out *stream, const void *buffer,
> + size_t bytes)
> +{
> + /* write data */
> +
> + return bytes;
> +}
> +
> +static uint32_t out_get_sample_rate(const struct audio_stream *stream)
> +{
> + struct hsp_stream_out *out = (struct hsp_stream_out *) stream;
> +
> + DBG("rate %u", out->cfg.rate);
> +
> + return out->cfg.rate;
> +}
> +
> +static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate)
> +{
> + DBG("rate %u", rate);
> +
> + return 0;
> +}
> +
> +static size_t out_get_buffer_size(const struct audio_stream *stream)
> +{
> + DBG("buf size %u", OUT_BUFFER_SIZE);
> +
> + return OUT_BUFFER_SIZE;
> +}
> +
> +static uint32_t out_get_channels(const struct audio_stream *stream)
> +{
> + DBG("");
> +
> + /* AudioFlinger can only provide stereo stream, so we return it here and
> + * later we'll downmix this to mono in case codec requires it
> + */
> + return AUDIO_CHANNEL_OUT_STEREO;
> +}
> +
> +static audio_format_t out_get_format(const struct audio_stream *stream)
> +{
> + struct hsp_stream_out *out = (struct hsp_stream_out *) stream;
> +
> + DBG("");
> +
> + return out->cfg.format;
> +}
> +
> +static int out_set_format(struct audio_stream *stream, audio_format_t format)
> +{
> + DBG("");
> +
> + return -ENOSYS;
> +}
> +
> +static int out_standby(struct audio_stream *stream)
> +{
> + DBG("");
> +
> + return 0;
> +}
> +
> +static int out_dump(const struct audio_stream *stream, int fd)
> +{
> + DBG("");
> +
> + return -ENOSYS;
> +}
> +
> +static int out_set_parameters(struct audio_stream *stream, const char *kvpairs)
> +{
> + DBG("%s", kvpairs);
> +
> + return 0;
> +}
> +
> +static char *out_get_parameters(const struct audio_stream *stream,
> + const char *keys)
> +{
> + DBG("");
> +
> + return strdup("");
> +}
> +
> +static uint32_t out_get_latency(const struct audio_stream_out *stream)
> +{
> + DBG("");
> +
> + return 0;
> +}
> +
> +static int out_set_volume(struct audio_stream_out *stream, float left,
> + float right)
> +{
> + DBG("");
> +
> + return -ENOSYS;
> +}
> +
> +static int out_get_render_position(const struct audio_stream_out *stream,
> + uint32_t *dsp_frames)
> +{
> + DBG("");
> +
> + return -ENOSYS;
> +}
> +
> +static int out_add_audio_effect(const struct audio_stream *stream,
> + effect_handle_t effect)
> +{
> + DBG("");
> +
> + return -ENOSYS;
> +}
> +
> +static int out_remove_audio_effect(const struct audio_stream *stream,
> + effect_handle_t effect)
> +{
> + DBG("");
> +
> + return -ENOSYS;
> +}
> +
> static int audio_open_output_stream(struct audio_hw_device *dev,
> audio_io_handle_t handle,
> audio_devices_t devices,
> audio_output_flags_t flags,
> struct audio_config *config,
> struct audio_stream_out **stream_out)
> -
> {
> + struct hsp_audio_dev *adev = (struct hsp_audio_dev *) dev;
> + struct hsp_stream_out *out;
> +
> DBG("");
>
> - return -EINVAL;
> + out = calloc(1, sizeof(struct hsp_stream_out));
> + if (!out)
> + return -ENOMEM;
> +
> + out->stream.common.get_sample_rate = out_get_sample_rate;
> + out->stream.common.set_sample_rate = out_set_sample_rate;
> + out->stream.common.get_buffer_size = out_get_buffer_size;
> + out->stream.common.get_channels = out_get_channels;
> + out->stream.common.get_format = out_get_format;
> + out->stream.common.set_format = out_set_format;
> + out->stream.common.standby = out_standby;
> + out->stream.common.dump = out_dump;
> + out->stream.common.set_parameters = out_set_parameters;
> + out->stream.common.get_parameters = out_get_parameters;
> + out->stream.common.add_audio_effect = out_add_audio_effect;
> + out->stream.common.remove_audio_effect = out_remove_audio_effect;
> + out->stream.get_latency = out_get_latency;
> + out->stream.set_volume = out_set_volume;
> + out->stream.write = out_write;
> + out->stream.get_render_position = out_get_render_position;
> +
> + out->cfg.format = AUDIO_STREAM_DEFAULT_FORMAT;
> + out->cfg.channels = AUDIO_CHANNEL_OUT_MONO;
> + out->cfg.rate = AUDIO_STREAM_DEFAULT_RATE;
> +
> + *stream_out = &out->stream;
> + adev->out = out;
> +
> + return 0;
> }
>
> static void audio_close_output_stream(struct audio_hw_device *dev,
> --
> 1.8.3.2
>
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--
Luiz Augusto von Dentz
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